[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 [A