[Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> INVITE sip:222@10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK5RO86S6mxetv0ZZX Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 49736409 39387904 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: --- (12 headers 9 lines) --- [Oct 4 10:52:54] DEBUG[7451] acl.c: For destination '10.10.10.2', our source address is '10.10.10.103'. [Oct 4 10:52:54] DEBUG[7451] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:52:54] DEBUG[7451] chan_sip.c: Allocating new SIP dialog for anKGyr89UvIu0Ac5@10.10.10.2 - INVITE (No RTP) [Oct 4 10:52:54] DEBUG[7451][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces" [Oct 4 10:52:54] DEBUG[7451][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 4 10:52:54] DEBUG[7451][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Using INVITE request as basis request - anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found peer 'alice' for 'alice' from 10.10.10.2:5060 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK5RO86S6mxetv0ZZX;received=10.10.10.2 From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as2be08018 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 1 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="739a9e61" Content-Length: 0 <------------> [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'anKGyr89UvIu0Ac5@10.10.10.2' in 6400 ms (Method: INVITE) [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> ACK sip:222@10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK5RO86S6mxetv0ZZX Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as2be08018 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 Contact: CSeq: 1 ACK Content-Length: 0 <-------------> [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: --- (10 headers 0 lines) --- [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Stopping retransmission on 'anKGyr89UvIu0Ac5@10.10.10.2' of Response 1: Match Found [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> INVITE sip:222@10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKv8Z1Mj5isnMaHtLT Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="739a9e61", uri="sip:222@10.10.10.103", response="626bf276fc1b554c52cda9c1885e9d97", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 19916583 61523009 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: --- (13 headers 9 lines) --- [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Using INVITE request as basis request - anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found peer 'alice' for 'alice' from 10.10.10.2:5060 [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fa2c4002430' [Oct 4 10:52:54] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Allocated port 15398 for RTP instance '0x7fa2c4002430' [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: RTP instance '0x7fa2c4002430' is setup and ready to go [Oct 4 10:52:54] DEBUG[7451][C-00000000] acl.c: Multiple addresses. Using the first only [Oct 4 10:52:54] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fa2c4002430' [Oct 4 10:52:54] VERBOSE[7451][C-00000000] netsock2.c: Using SIP RTP CoS mark 5 [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP o=- 19916583 61523009 IN IP4 10.10.10.2... OK. [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP s=SIP CALL... UNSUPPORTED OR FAILED. [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.2... OK. [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 8 [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7fa27b672470 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 101 [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7fa27b672470 [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 4 10:52:54] DEBUG[7451][C-00000000] acl.c: For destination '10.10.10.2', our source address is '10.10.10.103'. [Oct 4 10:52:54] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fa2c4002430' [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Peer audio RTP is at port 10.10.10.2:1780 [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 8 (0x7fa2c4009398) from 0x7fa27b672470 to 0x7fa2c40025f8 [Oct 4 10:52:54] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 101 (0x7fa2c4005ec8) from 0x7fa27b672470 to 0x7fa2c40025f8 [Oct 4 10:52:54] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fa2c4002430' [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Checking SIP call limits for device alice [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Updating call counter for incoming call [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: Looking for 222 in local (domain 10.10.10.103) [Oct 4 10:52:54] DEBUG[7410] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 4 10:52:54] DEBUG[7451][C-00000000] channel.c: Channel 0x7fa2c4011f00 'SIP/alice-00000000' allocated [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: *** Our native formats are (alaw) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: *** Joint capabilities are (alaw) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263) [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: This channel will not be able to handle video. [Oct 4 10:52:54] VERBOSE[7451][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: SIP/alice-00000000: New call is still down.... Trying... [Oct 4 10:52:54] VERBOSE[7451][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKv8Z1Mj5isnMaHtLT;received=10.10.10.2 From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:52:54] DEBUG[7421] chan_sip.c: Checking device state for peer alice [Oct 4 10:52:54] DEBUG[7421] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx.c: Launching 'Dial' [Oct 4 10:52:54] VERBOSE[7487][C-00000000] pbx.c: Executing [222@local:1] Dial("SIP/alice-00000000", "SIP/bob") in new stack [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up 222@local:-1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up 222@default:-1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up 222@demo:-1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Allocating new SIP dialog for 51fb28330de368c317b8a8d17ffc0332@127.0.1.1:5060 - INVITE (No RTP) [Oct 4 10:52:54] DEBUG[7487][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fa2cc016730' [Oct 4 10:52:54] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Allocated port 11478 for RTP instance '0x7fa2cc016730' [Oct 4 10:52:54] DEBUG[7487][C-00000000] rtp_engine.c: RTP instance '0x7fa2cc016730' is setup and ready to go [Oct 4 10:52:54] DEBUG[7487][C-00000000] acl.c: Multiple addresses. Using the first only [Oct 4 10:52:54] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fa2cc016730' [Oct 4 10:52:54] VERBOSE[7487][C-00000000] netsock2.c: Using SIP RTP CoS mark 5 [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 4 10:52:54] DEBUG[7487][C-00000000] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: SIP call-id changed from '51fb28330de368c317b8a8d17ffc0332@127.0.1.1:5060' to '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' [Oct 4 10:52:54] DEBUG[7487][C-00000000] channel.c: Channel 0x7fa2cc0198b0 'SIP/bob-00000001' allocated [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: *** Our native formats are (alaw) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: *** Joint capabilities are (alaw) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: This channel will not be able to handle video. [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up @local:1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up @default:1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up @demo:1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] pbx_lua.c: Looking up @demo:1 [Oct 4 10:52:54] DEBUG[7487][C-00000000] channel_internal_api.c: Channel Call ID changing from [C-00000000] to [C-00000000] [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Outgoing Call for bob [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Updating call counter for outgoing call [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: This call needs video offers, but there's no video support enabled! [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: ** Our capability: (alaw|ulaw|gsm|h263) Video flag: False Text flag: False [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: ** Our prefcodec: (alaw) [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Audio is at 11478 [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Adding codec alaw to SDP [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Adding codec ulaw to SDP [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Adding codec gsm to SDP [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw|gsm|h263) [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:52:54] VERBOSE[7487][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.100:5060: INVITE sip:bob@10.10.10.100;line=60c78014ea2900f SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK3e2d60b0 Max-Forwards: 70 From: "alice" ;tag=as6d43b9aa To: Contact: Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Date: Wed, 04 Oct 2017 08:52:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 v=0 o=root 2067263268 2067263268 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 11478 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 4 10:52:54] DEBUG[7487][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:52:54] VERBOSE[7487][C-00000000] app_dial.c: Called SIP/bob [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK3e2d60b0 From: "alice" ;tag=as6d43b9aa To: Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 102 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 4 10:52:54] VERBOSE[7451] chan_sip.c: --- (8 headers 0 lines) --- [Oct 4 10:52:54] DEBUG[7451][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' Request 102: Found [Oct 4 10:52:55] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK3e2d60b0 From: "alice" ;tag=as6d43b9aa To: ;tag=646368124 Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 4 10:52:55] VERBOSE[7451] chan_sip.c: --- (9 headers 0 lines) --- [Oct 4 10:52:55] DEBUG[7451][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' Request 102: Found [Oct 4 10:52:55] VERBOSE[7451][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Oct 4 10:52:55] DEBUG[7421] chan_sip.c: Checking device state for peer bob [Oct 4 10:52:55] DEBUG[7421] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 4 10:52:55] VERBOSE[7487][C-00000000] app_dial.c: SIP/bob-00000001 is ringing [Oct 4 10:52:55] DEBUG[7487][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/alice-00000000' with that of 'SIP/bob-00000001' [Oct 4 10:52:55] VERBOSE[7487][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKv8Z1Mj5isnMaHtLT;received=10.10.10.2 From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as043f2133 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 4 10:52:55] DEBUG[7487][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK3e2d60b0 From: "alice" ;tag=as6d43b9aa To: ;tag=646368124 Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 182 v=0 o=bob 1964 1318 IN IP4 10.10.10.100 s=Talk c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: --- (10 headers 9 lines) --- [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Acked pending invite 102 [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Stopping retransmission on '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' of Request 102: Match Found [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP o=bob 1964 1318 IN IP4 10.10.10.100... OK. [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.100... OK. [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 8 [Oct 4 10:53:02] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7fa27b6716f0 [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 101 [Oct 4 10:53:02] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7fa27b6716f0 [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 4 10:53:02] DEBUG[7451][C-00000000] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 4 10:53:02] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fa2cc016730' [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 4 10:53:02] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 8 (0x7fa2c4003b98) from 0x7fa27b6716f0 to 0x7fa2cc0168f8 [Oct 4 10:53:02] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 101 (0x7fa2c4004008) from 0x7fa27b6716f0 to 0x7fa2cc0168f8 [Oct 4 10:53:02] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fa2cc016730' [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Updating call counter for outgoing call [Oct 4 10:53:02] VERBOSE[7451][C-00000000] sip/route.c: sip_route_dump: route/path hop: [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Strict routing enforced for session 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: set_destination: set destination to 10.10.10.100:5060 [Oct 4 10:53:02] VERBOSE[7451][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.10.10.100:5060: ACK sip:vita@10.10.10.100 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK3bdaa9a3 Max-Forwards: 70 From: "alice" ;tag=as6d43b9aa To: ;tag=646368124 Contact: Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'ACK sip:vit' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:02] VERBOSE[7487][C-00000000] app_dial.c: SIP/bob-00000001 answered SIP/alice-00000000 [Oct 4 10:53:02] DEBUG[7487][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/alice-00000000' with that of 'SIP/bob-00000001' [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: SIP answering channel: SIP/alice-00000000 [Oct 4 10:53:02] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 4 10:53:02] DEBUG[7410] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Oct 4 10:53:02] VERBOSE[7487][C-00000000] chan_sip.c: Audio is at 15398 [Oct 4 10:53:02] VERBOSE[7487][C-00000000] chan_sip.c: Adding codec alaw to SDP [Oct 4 10:53:02] VERBOSE[7487][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 4 10:53:02] DEBUG[7421] chan_sip.c: Checking device state for peer bob [Oct 4 10:53:02] DEBUG[7421] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 4 10:53:02] VERBOSE[7487][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKv8Z1Mj5isnMaHtLT;received=10.10.10.2 From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as043f2133 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 232117809 232117809 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 15398 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Oct 4 10:53:02] DEBUG[7487][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:02] DEBUG[7421] chan_sip.c: Checking device state for peer alice [Oct 4 10:53:02] DEBUG[7421] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07' can not use native RTP bridge as two channels are required [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Chose bridge technology simple_bridge [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling simple_bridge technology constructor [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling simple_bridge technology start [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc028d50(SIP/bob-00000001) is joining [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: pushing 0x7fa2cc028d50(SIP/bob-00000001) [Oct 4 10:53:02] VERBOSE[7490][C-00000000] bridge_channel.c: Channel SIP/bob-00000001 joined 'simple_bridge' basic-bridge <5c2f1221-f587-4b9e-8638-5657d4491a07> [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07' can not use native RTP bridge as two channels are required [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Chose bridge technology simple_bridge [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07 is already using the new technology. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc028d50(SIP/bob-00000001) is joining simple_bridge technology [Oct 4 10:53:02] DEBUG[7490][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc01e3f0(SIP/alice-00000000) is joining [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: pushing 0x7fa2cc01e3f0(SIP/alice-00000000) [Oct 4 10:53:02] VERBOSE[7487][C-00000000] bridge_channel.c: Channel SIP/alice-00000000 joined 'simple_bridge' basic-bridge <5c2f1221-f587-4b9e-8638-5657d4491a07> [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Checking compatability for channels 'SIP/bob-00000001' and 'SIP/alice-00000000' [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Chose bridge technology native_rtp [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling native_rtp technology constructor [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: moving 0x7fa2cc028d50(SIP/bob-00000001) to dummy bridge temporarily [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc028d50(SIP/bob-00000001) is leaving simple_bridge technology (dummy) [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling simple_bridge technology stop [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc01e3f0(SIP/alice-00000000) is joining native_rtp technology [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Channel 'SIP/alice-00000000' is joining bridge tech [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Attaching hook data 0x7fa2cc023c70 to 'SIP/alice-00000000' [Oct 4 10:53:02] DEBUG[7422] cdr.c: Finalized CDR for SIP/bob-00000001 - start 1507107174.917451 answer 1507107182.300152 end 1507107182.302212 dispo ANSWERED [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc028d50(SIP/bob-00000001) is joining native_rtp technology [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Channel 'SIP/bob-00000001' is joining bridge tech [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Attaching hook data 0x7fa2cc0296d0 to 'SIP/bob-00000001' [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech starting 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'none' [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling native_rtp technology start [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling simple_bridge technology destructor [Oct 4 10:53:02] DEBUG[7410] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 4 10:53:02] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Checking compatability for channels 'SIP/alice-00000000' and 'SIP/bob-00000001' [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Chose bridge technology native_rtp [Oct 4 10:53:02] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07 is already using the new technology. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Checking compatability for channels 'SIP/alice-00000000' and 'SIP/bob-00000001' [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Chose bridge technology native_rtp [Oct 4 10:53:02] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07 is already using the new technology. [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: Retransmitting #1 (no NAT) to 10.10.10.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKv8Z1Mj5isnMaHtLT;received=10.10.10.2 From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as043f2133 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 232117809 232117809 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 15398 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 4 10:53:02] DEBUG[7451] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> ACK sip:222@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKJiaLvbypkIKFXcfq Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as043f2133 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="739a9e61", uri="sip:222@10.10.10.103:5060", response="aabdfbcf12dbf5da1266709680c16b1b", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 <-------------> [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: --- (11 headers 0 lines) --- [Oct 4 10:53:02] DEBUG[7451][C-00000000] chan_sip.c: Stopping retransmission on 'anKGyr89UvIu0Ac5@10.10.10.2' of Response 2: Match Found [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> ACK sip:222@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKJiaLvbypkIKFXcfq Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=wdCoyriZfeOAucrp To: "222" ;tag=as043f2133 Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="739a9e61", uri="sip:222@10.10.10.103:5060", response="aabdfbcf12dbf5da1266709680c16b1b", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 <-------------> [Oct 4 10:53:02] VERBOSE[7451] chan_sip.c: --- (11 headers 0 lines) --- [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> REGISTER sip:10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK3qslbVGjWnh4xeJh Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=969RNleaZAcgL9MD To: "alice" Call-ID: r3Y8nxCW8pgJq6Lj@10.10.10.2 CSeq: 15386 REGISTER Contact: Expires: 60 Content-Length: 0 <-------------> [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: --- (11 headers 0 lines) --- [Oct 4 10:53:03] DEBUG[7451] acl.c: For destination '10.10.10.2', our source address is '10.10.10.103'. [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Allocating new SIP dialog for r3Y8nxCW8pgJq6Lj@10.10.10.2 - REGISTER (No RTP) [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK3qslbVGjWnh4xeJh;received=10.10.10.2 From: "alice" ;tag=969RNleaZAcgL9MD To: "alice" ;tag=as733393e7 Call-ID: r3Y8nxCW8pgJq6Lj@10.10.10.2 CSeq: 15386 REGISTER Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4258acea" Content-Length: 0 <------------> [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Scheduling destruction of SIP dialog 'r3Y8nxCW8pgJq6Lj@10.10.10.2' in 32000 ms (Method: REGISTER) [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> REGISTER sip:10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKinXkqrKfZeLE9x0C Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=969RNleaZAcgL9MD To: "alice" Call-ID: r3Y8nxCW8pgJq6Lj@10.10.10.2 CSeq: 15387 REGISTER Contact: Expires: 60 Authorization: Digest username="alice", realm="asterisk", nonce="4258acea", uri="sip:10.10.10.103", response="bab650a0e6c3fb42a286832b657ea361", algorithm=MD5 Content-Length: 0 <-------------> [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: --- (12 headers 0 lines) --- [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: build_path: do not use Path headers [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Allocating new SIP dialog for 45cab1562b5ebb274cc0f4ef3d3be7c7@127.0.1.1:5060 - OPTIONS (No RTP) [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 4 10:53:03] DEBUG[7451] acl.c: For destination '10.10.10.2', our source address is '10.10.10.103'. [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: SIP call-id changed from '45cab1562b5ebb274cc0f4ef3d3be7c7@127.0.1.1:5060' to '35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060' [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Initializing initreq for method OPTIONS - callid 35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: OPTIONS sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK04543519 Max-Forwards: 70 From: "asterisk" ;tag=as5d4ddfdd To: Contact: Call-ID: 35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Date: Wed, 04 Oct 2017 08:53:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKinXkqrKfZeLE9x0C;received=10.10.10.2 From: "alice" ;tag=969RNleaZAcgL9MD To: "alice" ;tag=as733393e7 Call-ID: r3Y8nxCW8pgJq6Lj@10.10.10.2 CSeq: 15387 REGISTER Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Wed, 04 Oct 2017 08:53:03 GMT Content-Length: 0 <------------> [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Scheduling destruction of SIP dialog 'r3Y8nxCW8pgJq6Lj@10.10.10.2' in 32000 ms (Method: REGISTER) [Oct 4 10:53:03] DEBUG[7421] chan_sip.c: Checking device state for peer alice [Oct 4 10:53:03] DEBUG[7421] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK04543519 Call-ID: 35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060 CSeq: 102 OPTIONS From: "asterisk" ;tag=as5d4ddfdd To: ;tag=cWq4HtJRrGLPzLsg Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Accept: application/sdp, message/sipfrag, application/dtmf-relay Supported: replaces Content-Length: 0 <-------------> [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: --- (11 headers 0 lines) --- [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Stopping retransmission on '35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060' of Request 102: Match Found [Oct 4 10:53:03] DEBUG[7451] chan_sip.c: Destroying SIP dialog 35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060 [Oct 4 10:53:03] VERBOSE[7451] chan_sip.c: Really destroying SIP dialog '35f699cf4fb7495b15d5b2457ebb68a3@10.10.10.103:5060' Method: OPTIONS [Oct 4 10:53:04] DEBUG[7490][C-00000000] res_rtp_asterisk.c: Got RTCP report of 100 bytes from 10.10.10.100:7079 [Oct 4 10:53:07] DEBUG[7490][C-00000000] res_rtp_asterisk.c: Got RTCP report of 100 bytes from 10.10.10.100:7079 [Oct 4 10:53:07] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Got RTCP report of 60 bytes from 10.10.10.2:1781 [Oct 4 10:53:07] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> INVITE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK1221805975 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 2 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Call on hold Content-Length: 212 v=0 o=bob 1964 1319 IN IP4 10.10.10.100 s=Talk c=IN IP4 0.0.0.0 t=0 0 m=audio 7078 RTP/AVP 8 101 c=IN IP4 10.10.10.100 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly <-------------> [Oct 4 10:53:07] VERBOSE[7451] chan_sip.c: --- (13 headers 11 lines) --- [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP o=bob 1964 1319 IN IP4 10.10.10.100... OK. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 0.0.0.0... OK. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 8 [Oct 4 10:53:07] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7fa27b672470 [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 101 [Oct 4 10:53:07] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7fa27b672470 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.10.100... OK. [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 4 10:53:07] DEBUG[7451][C-00000000] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 4 10:53:07] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fa2cc016730' [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 4 10:53:07] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 8 (0x7fa2c4010b68) from 0x7fa27b672470 to 0x7fa2cc0168f8 [Oct 4 10:53:07] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 101 (0x7fa2c4011428) from 0x7fa27b672470 to 0x7fa2cc0168f8 [Oct 4 10:53:07] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fa2cc016730' [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Got a SIP re-invite for call 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: SIP/bob-00000001: This call is UP.... [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK1221805975;received=10.10.10.100;rport=5060 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: ** Our prefcodec: (alaw) [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Audio is at 11478 [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec alaw to SDP [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec ulaw to SDP [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec gsm to SDP [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 4 10:53:07] VERBOSE[7451][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK1221805975;received=10.10.10.100;rport=5060 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 2067263268 2067263269 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 11478 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=recvonly <------------> [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:07] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> jaK <-------------> [Oct 4 10:53:07] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech stopping 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'SIP/alice-00000000' [Oct 4 10:53:07] DEBUG[7487][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/alice-00000000' and 'SIP/bob-00000001' - media will flow through Asterisk core [Oct 4 10:53:07] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 4 10:53:07] VERBOSE[7487][C-00000000] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/alice-00000000' [Oct 4 10:53:07] DEBUG[7487][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 4 10:53:07] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 4 10:53:07] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech starting 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'SIP/alice-00000000' [Oct 4 10:53:07] DEBUG[7487][C-00000000] channel.c: Channel SIP/alice-00000000 setting write format path: slin -> alaw [Oct 4 10:53:07] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> ACK sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK506392828 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 4 10:53:07] VERBOSE[7451] chan_sip.c: --- (10 headers 0 lines) --- [Oct 4 10:53:07] DEBUG[7451][C-00000000] chan_sip.c: Stopping retransmission on '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' of Response 2: Match Found [Oct 4 10:53:07] DEBUG[7490][C-00000000] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 4 10:53:07] DEBUG[7490][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fa2cc016730' [Oct 4 10:53:07] DEBUG[7487][C-00000000] res_musiconhold.c: SIP/alice-00000000 Opened file 0 '/var/lib/asterisk/moh/macroform-the_simplicity' [Oct 4 10:53:07] DEBUG[7487][C-00000000] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Oct 4 10:53:10] DEBUG[7490][C-00000000] res_rtp_asterisk.c: Got RTCP report of 100 bytes from 10.10.10.100:7079 [Oct 4 10:53:11] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> INVITE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK478975983 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 3 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Call resuming Content-Length: 182 v=0 o=bob 1964 1320 IN IP4 10.10.10.100 s=Talk c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 4 10:53:11] VERBOSE[7451] chan_sip.c: --- (13 headers 9 lines) --- [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP o=bob 1964 1320 IN IP4 10.10.10.100... OK. [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.100... OK. [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 8 [Oct 4 10:53:11] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7fa27b672470 [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Found RTP audio format 101 [Oct 4 10:53:11] DEBUG[7451][C-00000000] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7fa27b672470 [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 4 10:53:11] DEBUG[7451][C-00000000] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 4 10:53:11] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fa2cc016730' [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 4 10:53:11] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 8 (0x7fa2c4016358) from 0x7fa27b672470 to 0x7fa2cc0168f8 [Oct 4 10:53:11] DEBUG[7451][C-00000000] rtp_engine.c: Copying payload 101 (0x7fa2c4001d68) from 0x7fa27b672470 to 0x7fa2cc0168f8 [Oct 4 10:53:11] DEBUG[7451][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fa2cc016730' [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Got a SIP re-invite for call 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: SIP/bob-00000001: This call is UP.... [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK478975983;received=10.10.10.100;rport=5060 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 3 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: ** Our prefcodec: (alaw) [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Audio is at 11478 [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec alaw to SDP [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec ulaw to SDP [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Adding codec gsm to SDP [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Setting framing on incoming call: 0 [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 4 10:53:11] VERBOSE[7451][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK478975983;received=10.10.10.100;rport=5060 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 3 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 2067263268 2067263270 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 11478 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:11] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech starting 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'SIP/alice-00000000' [Oct 4 10:53:11] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 4 10:53:11] VERBOSE[7487][C-00000000] res_musiconhold.c: Stopped music on hold on SIP/alice-00000000 [Oct 4 10:53:11] DEBUG[7487][C-00000000] channel.c: Channel SIP/alice-00000000 setting write format path: alaw -> alaw [Oct 4 10:53:11] DEBUG[7487][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 4 10:53:11] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 4 10:53:11] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech starting 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'SIP/alice-00000000' [Oct 4 10:53:11] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> ACK sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK11156413 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 3 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 4 10:53:11] VERBOSE[7451] chan_sip.c: --- (10 headers 0 lines) --- [Oct 4 10:53:11] DEBUG[7451][C-00000000] chan_sip.c: Stopping retransmission on '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' of Response 3: Match Found [Oct 4 10:53:12] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Got RTCP report of 60 bytes from 10.10.10.2:1781 [Oct 4 10:53:13] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> BYE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK1947262464 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 4 BYE Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 4 10:53:13] VERBOSE[7451] chan_sip.c: --- (10 headers 0 lines) --- [Oct 4 10:53:13] DEBUG[7451][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:13] VERBOSE[7451][C-00000000] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 4 10:53:13] DEBUG[7451][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:13] VERBOSE[7451][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060' in 6400 ms (Method: BYE) [Oct 4 10:53:13] DEBUG[7451][C-00000000] chan_sip.c: Received bye, issuing owner hangup [Oct 4 10:53:13] VERBOSE[7451][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK1947262464;received=10.10.10.100;rport=5060 From: ;tag=646368124 To: "alice" ;tag=as6d43b9aa Call-ID: 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 CSeq: 4 BYE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 4 10:53:13] DEBUG[7451][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_channel.c: Setting 0x7fa2cc028d50(SIP/bob-00000001) state from:0 to:1 [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: pulling 0x7fa2cc028d50(SIP/bob-00000001) [Oct 4 10:53:13] VERBOSE[7490][C-00000000] bridge_channel.c: Channel SIP/bob-00000001 left 'native_rtp' basic-bridge <5c2f1221-f587-4b9e-8638-5657d4491a07> [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc028d50(SIP/bob-00000001) is leaving native_rtp technology [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Channel 'SIP/bob-00000001' is leaving bridge tech [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Detaching hook data 0x7fa2cc029738 from 'SIP/bob-00000001' [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Tech stopping 'SIP/alice-00000000' and 'SIP/bob-00000001' with target 'none' [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/alice-00000000' and 'SIP/bob-00000001' - media will flow through Asterisk core [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_native_rtp.c: Destroying channel tech_pvt data 0x7fa2cc0296d0 [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: dissolving bridge with cause 16(Normal Clearing) [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge_channel.c: Setting 0x7fa2cc01e3f0(SIP/alice-00000000) state from:0 to:2 [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: queueing action type:13 sub:1001 [Oct 4 10:53:13] DEBUG[7422] cdr.c: Finalized CDR for SIP/alice-00000000 - start 1507107174.895616 answer 1507107182.300383 end 1507107193.843011 dispo ANSWERED [Oct 4 10:53:13] DEBUG[7490][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07 is dissolved, not performing smart bridge operation. [Oct 4 10:53:13] DEBUG[7490][C-00000000] channel.c: Channel 0x7fa2cc0198b0 'SIP/bob-00000001' hanging up. Refs: 2 [Oct 4 10:53:13] DEBUG[7490][C-00000000] chan_sip.c: Hangup call SIP/bob-00000001, SIP callid 7398d4eb117170610b9e01c953bbe27e@10.10.10.103:5060 [Oct 4 10:53:13] DEBUG[7490][C-00000000] channel.c: Channel 0x7fa2cc0198b0 'SIP/bob-00000001' destroying [Oct 4 10:53:13] DEBUG[7422] cdr.c: CDR for SIP/bob-00000001 is dialed and has no Party B; discarding [Oct 4 10:53:13] DEBUG[7421] chan_sip.c: Checking device state for peer bob [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: pulling 0x7fa2cc01e3f0(SIP/alice-00000000) [Oct 4 10:53:13] DEBUG[7421] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 4 10:53:13] VERBOSE[7487][C-00000000] bridge_channel.c: Channel SIP/alice-00000000 left 'native_rtp' basic-bridge <5c2f1221-f587-4b9e-8638-5657d4491a07> [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge_channel.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: 0x7fa2cc01e3f0(SIP/alice-00000000) is leaving native_rtp technology [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Channel 'SIP/alice-00000000' is leaving bridge tech [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge_native_rtp.c: Bridge '5c2f1221-f587-4b9e-8638-5657d4491a07'. Detaching hook data 0x7fa2cc023cd8 from 'SIP/alice-00000000' [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge_native_rtp.c: Destroying channel tech_pvt data 0x7fa2cc023c70 [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07 is dissolved, not performing smart bridge operation. [Oct 4 10:53:13] DEBUG[7487][C-00000000] res_rtp_asterisk.c: Changing ssrc from 1491666895 to 1901421351 due to a source change [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: actually destroying basic bridge, nobody wants it anymore [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling basic bridge destructor [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling native_rtp technology stop [Oct 4 10:53:13] DEBUG[7487][C-00000000] bridge.c: Bridge 5c2f1221-f587-4b9e-8638-5657d4491a07: calling native_rtp technology destructor [Oct 4 10:53:13] DEBUG[7487][C-00000000] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 4 10:53:13] DEBUG[7487][C-00000000] pbx.c: Spawn extension (local,222,1) exited non-zero on 'SIP/alice-00000000' [Oct 4 10:53:13] VERBOSE[7487][C-00000000] pbx.c: Spawn extension (local, 222, 1) exited non-zero on 'SIP/alice-00000000' [Oct 4 10:53:13] DEBUG[7487][C-00000000] channel.c: Soft-Hanging (0x10) up channel 'SIP/alice-00000000' [Oct 4 10:53:13] DEBUG[7487][C-00000000] pbx_lua.c: Looking up h@local:1 [Oct 4 10:53:13] DEBUG[7487][C-00000000] pbx_lua.c: Looking up h@default:1 [Oct 4 10:53:13] DEBUG[7487][C-00000000] pbx_lua.c: Looking up h@demo:1 [Oct 4 10:53:13] DEBUG[7487][C-00000000] pbx_lua.c: Looking up h@demo:1 [Oct 4 10:53:13] DEBUG[7487][C-00000000] channel.c: Channel 0x7fa2c4011f00 'SIP/alice-00000000' hanging up. Refs: 2 [Oct 4 10:53:13] DEBUG[7487][C-00000000] chan_sip.c: Hangup call SIP/alice-00000000, SIP callid anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:53:13] VERBOSE[7487][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'anKGyr89UvIu0Ac5@10.10.10.2' in 6400 ms (Method: ACK) [Oct 4 10:53:13] DEBUG[7487][C-00000000] chan_sip.c: Strict routing enforced for session anKGyr89UvIu0Ac5@10.10.10.2 [Oct 4 10:53:13] VERBOSE[7487][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 4 10:53:13] VERBOSE[7487][C-00000000] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 4 10:53:13] VERBOSE[7487][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: BYE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK387d248f Max-Forwards: 70 From: "222" ;tag=as043f2133 To: "alice" ;tag=wdCoyriZfeOAucrp Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 102 BYE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Proxy-Authorization: Digest username="alice", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.103", nonce="739a9e61", response="6a263b27f43d0f86fa4bd8072db5807b" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Oct 4 10:53:13] DEBUG[7487][C-00000000] chan_sip.c: Trying to put 'BYE sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 4 10:53:13] DEBUG[7487][C-00000000] channel.c: Channel 0x7fa2c4011f00 'SIP/alice-00000000' destroying [Oct 4 10:53:13] DEBUG[7421] chan_sip.c: Checking device state for peer alice [Oct 4 10:53:13] DEBUG[7421] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 4 10:53:13] DEBUG[7422] res_config_sqlite.c: About to query table structure: SELECT sql FROM sqlite_master WHERE type='table' AND tbl_name='ast_cdr' [Oct 4 10:53:13] DEBUG[7422] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"alice" ','alice','222','local','SIP/alice-00000000','SIP/bob-00000001','Dial','SIP/bob','2017-10-04 10:52:54','2017-10-04 10:53:02','2017-10-04 10:53:13','18','11','ANSWERED','DOCUMENTATION','1507107174.0') [Oct 4 10:53:13] VERBOSE[7451] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK387d248f Call-ID: anKGyr89UvIu0Ac5@10.10.10.2 CSeq: 102 BYE From: "222" ;tag=as043f2133 To: "alice" ;tag=wdCoyriZfeOAucrp Contact: Content-Length: 0 <------------->