[Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Starting thread for SSL server [Apr 3 18:51:29] DEBUG[24556] chan_sip.c: SIP SSL server timed out [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 0 [ 57]: INVITE sip:+87654321@10.62.150.68;user=phone SIP/2.0 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 1 [108]: FROM: "user";epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 2 [ 48]: TO: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 3 [ 18]: CSEQ: 22303 INVITE [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 7 [ 82]: CONTACT: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 8 [ 19]: CONTENT-LENGTH: 523 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 9 [ 17]: SUPPORTED: 100rel [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 10 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 11 [ 29]: CONTENT-TYPE: application/sdp [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 12 [ 10]: ALLOW: ACK [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 13 [ 37]: Allow: CANCEL,BYE,INVITE,PRACK,UPDATE [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 14 [ 0]: [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: <--- SIP read from TLS:10.62.150.23:62847 ---> INVITE sip:+87654321@10.62.150.68;user=phone SIP/2.0 FROM: "user";epid=89E6832516;tag=15165c8a26 TO: CSEQ: 22303 INVITE CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda CONTACT: CONTENT-LENGTH: 523 SUPPORTED: 100rel USER-AGENT: RTCC/4.0.0.0 MediationServer CONTENT-TYPE: application/sdp ALLOW: ACK Allow: CANCEL,BYE,INVITE,PRACK,UPDATE v=0 o=- 173 1 IN IP4 10.62.150.23 s=session c=IN IP4 10.62.150.23 b=CT:1000 t=0 0 m=audio 56268 RTP/SAVP 97 101 13 0 8 c=IN IP4 10.62.150.23 a=rtcp:56269 a=label:Audio a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HkrE2/vdmirPBL6v8pVf4Wp4VD/1W38KxqVpSLOx|2^31|1:1 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:seJsLO8PNmEEXYT4Aoc1Cuz3Y8ef9N6UEWzzAwdM|2^31 a=sendrecv a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 <-------------> [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 0 [ 57]: INVITE sip:+87654321@10.62.150.68;user=phone SIP/2.0 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 1 [108]: FROM: "user";epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 2 [ 48]: TO: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 3 [ 18]: CSEQ: 22303 INVITE [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 7 [ 82]: CONTACT: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 8 [ 19]: CONTENT-LENGTH: 523 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 9 [ 17]: SUPPORTED: 100rel [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 10 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 11 [ 29]: CONTENT-TYPE: application/sdp [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 12 [ 10]: ALLOW: ACK [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 13 [ 37]: Allow: CANCEL,BYE,INVITE,PRACK,UPDATE [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Header 14 [ 0]: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 0 [ 3]: v=0 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 1 [ 29]: o=- 173 1 IN IP4 10.62.150.23 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 2 [ 9]: s=session [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.62.150.23 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 4 [ 9]: b=CT:1000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 5 [ 5]: t=0 0 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 6 [ 36]: m=audio 56268 RTP/SAVP 97 101 13 0 8 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 7 [ 21]: c=IN IP4 10.62.150.23 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 8 [ 12]: a=rtcp:56269 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 9 [ 13]: a=label:Audio [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 10 [ 91]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HkrE2/vdmirPBL6v8pVf4Wp4VD/1W38KxqVpSLOx|2^31|1:1 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 11 [ 87]: a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:seJsLO8PNmEEXYT4Aoc1Cuz3Y8ef9N6UEWzzAwdM|2^31 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 12 [ 10]: a=sendrecv [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 13 [ 20]: a=rtpmap:97 RED/8000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-16 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 16 [ 19]: a=rtpmap:13 CN/8000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 17 [ 20]: a=rtpmap:0 PCMU/8000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 18 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Body 19 [ 10]: a=ptime:20 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: --- (14 headers 20 lines) --- [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: = Looking for Call ID: 67c95c9a-199f-4864-8722-c69b12389c7c (Checking From) --From tag 15165c8a26 --To-tag [Apr 3 18:51:29] DEBUG[24573] acl.c: For destination '10.62.150.23', our source address is '10.62.150.68'. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Setting SIP_TRANSPORT_TLS with address 10.62.150.68:5067 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Allocating new SIP dialog for 67c95c9a-199f-4864-8722-c69b12389c7c - INVITE (No RTP) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Apr 3 18:51:29] DEBUG[24573] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [Apr 3 18:51:29] DEBUG[24573] sip/reqresp_parser.c: Found SIP option: -100rel- [Apr 3 18:51:29] DEBUG[24573] sip/reqresp_parser.c: Matched SIP option: 100rel [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting '10.62.150.23:62847' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host '10.62.150.23' and port '62847'. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Sending to 10.62.150.23:62847 (no NAT) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Initializing initreq for method INVITE - callid 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Using INVITE request as basis request - 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting 'ocsdom03.ngnocs.local' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host 'ocsdom03.ngnocs.local' and port ''. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found peer 'pbx' for '+12345678' from 10.62.150.23:62847 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x8cb22f4' [Apr 3 18:51:29] DEBUG[24573] res_rtp_asterisk.c: Allocated port 8282 for RTP instance '0x8cb22f4' [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: RTP instance '0x8cb22f4' is setup and ready to go [Apr 3 18:51:29] DEBUG[24573] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8cb22f4' [Apr 3 18:51:29] VERBOSE[24573] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Setting NAT on RTP to Off [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP o=- 173 1 IN IP4 10.62.150.23... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting '10.62.150.23' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host '10.62.150.23' and port ''. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP c=IN IP4 10.62.150.23... OK. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP b=CT:1000... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found RTP audio format 97 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Setting payload 97 based on m type on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found RTP audio format 101 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Setting payload 101 based on m type on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found RTP audio format 13 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Setting payload 13 based on m type on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found RTP audio format 0 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Setting payload 0 based on m type on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found RTP audio format 8 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Setting payload 8 based on m type on 0xb6ac6e98 [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting '10.62.150.23' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host '10.62.150.23' and port ''. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.62.150.23... OK. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtcp:56269... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=label:Audio... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] sip/sdp_crypto.c: local_key64 ldD7DqsCBntYJxxgX/2S4xvpScleKz7EYOZzXTXB len 40 [Apr 3 18:51:29] DEBUG[24573] res_srtp.c: Adding new policy for SSRC 1625075369 [Apr 3 18:51:29] DEBUG[24573] sip/sdp_crypto.c: SRTP policy activated [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HkrE2/vdmirPBL6v8pVf4Wp4VD/1W38KxqVpSLOx|2^31|1:1... OK. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: We've already processed a crypto attribute, skipping 'crypto:2 AES_CM_128_HMAC_SHA1_80 inline:seJsLO8PNmEEXYT4Aoc1Cuz3Y8ef9N6UEWzzAwdM|2^31' [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:seJsLO8PNmEEXYT4Aoc1Cuz3Y8ef9N6UEWzzAwdM|2^31... UNSUPPORTED. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Unsetting payload 97 on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found unknown media description format RED for ID 97 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 RED/8000... UNSUPPORTED. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found audio description format CN for ID 13 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found audio description format PCMU for ID 0 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Found audio description format PCMA for ID 8 [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Incorporating payload 0 on 0xb6ac6e98 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Incorporating payload 8 on 0xb6ac6e98 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Incorporating payload 13 on 0xb6ac6e98 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Incorporating payload 101 on 0xb6ac6e98 [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Capabilities: us - (alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) [Apr 3 18:51:29] DEBUG[24573] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8cb22f4' [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Peer audio RTP is at port 10.62.150.23:56268 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Copying payload 0 from 0xb6ac6e98 to 0x8cb24a0 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Copying payload 8 from 0xb6ac6e98 to 0x8cb24a0 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Copying payload 13 from 0xb6ac6e98 to 0x8cb24a0 [Apr 3 18:51:29] DEBUG[24573] rtp_engine.c: Copying payload 101 from 0xb6ac6e98 to 0x8cb24a0 [Apr 3 18:51:29] DEBUG[24573] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x8cb22f4' [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: We're settling with these formats: (alaw) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Checking SIP call limits for device [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Updating call counter for incoming call [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting '10.62.150.68' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host '10.62.150.68' and port ''. [Apr 3 18:51:29] DEBUG[24573] netsock2.c: Splitting 'ocsdom03.ngnocs.local' into... [Apr 3 18:51:29] DEBUG[24573] netsock2.c: ...host 'ocsdom03.ngnocs.local' and port ''. [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: Looking for +87654321 in from-pbx (domain 10.62.150.68) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: *** Our native formats are (alaw) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: *** Joint capabilities are (alaw) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: *** Our capabilities are (alaw) [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: This channel will not be able to handle video. [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: build_route: Contact hop: [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: list_route: hop: [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: SIP/pbx-00000003: New call is still down.... Trying... [Apr 3 18:51:29] VERBOSE[24573] chan_sip.c: <--- Transmitting (no NAT) to 10.62.150.23:62847 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda;received=10.62.150.23 From: "user";epid=89E6832516;tag=15165c8a26 To: Call-ID: 67c95c9a-199f-4864-8722-c69b12389c7c CSeq: 22303 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 3 18:51:29] DEBUG[24573] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 10.62.150.23:62847 [Apr 3 18:51:29] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - pbx [Apr 3 18:51:29] DEBUG[15607] chan_sip.c: Checking device state for peer pbx [Apr 3 18:51:29] DEBUG[15607] devicestate.c: Changing state for SIP/pbx - state 1 (Not in use) [Apr 3 18:51:29] DEBUG[15607] devicestate.c: device 'SIP/pbx' state '1' [Apr 3 18:51:29] DEBUG[24574] logger.c: CALL_ID [C-00000002] created by thread. [Apr 3 18:51:29] DEBUG[24574][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Apr 3 18:51:29] DEBUG[24574][C-00000002] pbx.c: Launching 'Progress' [Apr 3 18:51:29] VERBOSE[24574][C-00000002] pbx.c: -- Executing [+87654321@from-pbx:1] Progress("SIP/pbx-00000003", "") in new stack [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Setting framing from config on incoming call [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Audio is at 8282 [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: <--- Transmitting (no NAT) to 10.62.150.23:62847 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda;received=10.62.150.23 From: "user";epid=89E6832516;tag=15165c8a26 To: ;tag=as235f078c Call-ID: 67c95c9a-199f-4864-8722-c69b12389c7c CSeq: 22303 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 346 v=0 o=TELEflash.com 346530056 346530056 IN IP4 10.62.150.68 s=pbx session c=IN IP4 10.62.150.68 t=0 0 m=audio 8282 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ldD7DqsCBntYJxxgX/2S4xvpScleKz7EYOZzXTXB <------------> [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 183' onto TLS socket destined for 10.62.150.23:62847 [Apr 3 18:51:29] DEBUG[24574][C-00000002] pbx.c: Result of 'EXTEN' is '+87654321' [Apr 3 18:51:29] DEBUG[24574][C-00000002] pbx.c: Launching 'Dial' [Apr 3 18:51:29] VERBOSE[24574][C-00000002] pbx.c: -- Executing [+87654321@from-pbx:2] Dial("SIP/pbx-00000003", "SIP/+87654321@tas") in new stack [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Allocating new SIP dialog for 32fb414e1faf75500d9566e86ddbed42@127.0.0.1:5060 - INVITE (No RTP) [Apr 3 18:51:29] DEBUG[24574][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x8cc7d9c' [Apr 3 18:51:29] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Allocated port 20252 for RTP instance '0x8cc7d9c' [Apr 3 18:51:29] DEBUG[24574][C-00000002] rtp_engine.c: RTP instance '0x8cc7d9c' is setup and ready to go [Apr 3 18:51:29] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8cc7d9c' [Apr 3 18:51:29] VERBOSE[24574][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Apr 3 18:51:29] DEBUG[24574][C-00000002] acl.c: For destination '217.7.75.80', our source address is '10.62.150.68'. [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.62.150.68:5060 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** Our native formats are (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** Joint capabilities are (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** Our capabilities are (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: This channel will not be able to handle video. [Apr 3 18:51:29] DEBUG[24574][C-00000002] rtp_engine.c: Seeded SDP of 'SIP/tas-00000004' with that of 'SIP/pbx-00000003' [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable DIALEDTIME. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable ANSWEREDTIME. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable DIALEDPEERNAME. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable DIALEDPEERNUMBER. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable DIALSTATUS. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable SIPCALLID. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable SIPDOMAIN. [Apr 3 18:51:29] DEBUG[24574][C-00000002] channel.c: Not copying variable SIPURI. [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Outgoing Call for +87654321 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Updating call counter for outgoing call [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: ** Our prefcodec: (alaw) [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Audio is at 20252 [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 0 [ 45]: INVITE sip:+87654321@test.local SIP/2.0 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK1c477ddd [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 4 [ 36]: To: [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 5 [ 47]: Contact: [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 6 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 8 [ 35]: Date: Tue, 03 Apr 2012 16:51:29 GMT [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Apr 3 18:51:29] VERBOSE[24574][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 217.7.75.80:5060: INVITE sip:+87654321@test.local SIP/2.0 Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK1c477ddd Max-Forwards: 70 From: "user" ;tag=as1be54ed8 To: Contact: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 102 INVITE Date: Tue, 03 Apr 2012 16:51:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=session.com 621634435 621634435 IN IP4 10.62.150.68 s=pbx session c=IN IP4 10.62.150.68 t=0 0 m=audio 20252 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22179 [Apr 3 18:51:29] DEBUG[24574][C-00000002] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:29] VERBOSE[24574][C-00000002] app_dial.c: -- Called SIP/+87654321@tas [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c477ddd To: From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 102 INVITE Content-Length: 0 <-------------> [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c477ddd [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 2 [ 36]: To: [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: --- (7 headers 0 lines) --- [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: *** SIP TIMER: Cancelling retransmission #22179 - INVITE (got response) [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '77a598877fb2fdad09dc26f039c8facc@test.local' Request 102: Found [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: SIP response 100 to standard invite [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c477ddd To: ;tag=361f3a25 From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: CSeq: 102 INVITE WWW-Authenticate: Digest algorithm=MD5, nonce="2b5e06e094747daf2b5e06e0f11506e0b026768435f3086660000a7f39479e55e8e0bdbb", realm="test.local" Reason: TSSI;cause=401;reason=4010001 Content-Length: 0 <-------------> [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c477ddd [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 2 [ 49]: To: ;tag=361f3a25 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 5 [ 39]: Contact: [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 7 [141]: WWW-Authenticate: Digest algorithm=MD5, nonce="2b5e06e094747daf2b5e06e0f11506e0b026768435f3086660000a7f39479e55e8e0bdbb", realm="test.local" [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 8 [ 37]: Reason: TSSI;cause=401;reason=4010001 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: --- (10 headers 0 lines) --- [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag 361f3a25 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Acked pending invite 102 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Stopping retransmission on '77a598877fb2fdad09dc26f039c8facc@test.local' of Request 102: Match Found [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: SIP response 401 to standard invite [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: set_destination: Parsing for address/port to send to [Apr 3 18:51:29] DEBUG[15615] netsock2.c: Splitting 'test.local' into... [Apr 3 18:51:29] DEBUG[15615] netsock2.c: ...host 'test.local' and port ''. [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: set_destination: set destination to 217.7.75.80:5060 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: Transmitting (no NAT) to 217.7.75.80:5060: ACK sip:+87654321@test.local SIP/2.0 Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK1c477ddd Max-Forwards: 70 From: "user" ;tag=as1be54ed8 To: ;tag=361f3a25 Contact: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 102 ACK Content-Length: 0 --- [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Trying to put 'ACK sip:+49' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Auth attempt 1 on INVITE [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: ** Our prefcodec: (alaw) [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: Audio is at 20252 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: Adding codec 100004 (alaw) to SDP [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: Reliably Transmitting (no NAT) to 217.7.75.80:5060: INVITE sip:+87654321@test.local SIP/2.0 Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK46d63744 Max-Forwards: 70 From: "user" ;tag=as1be54ed8 To: Contact: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 103 INVITE Authorization: Digest username="fmc", realm="test.local", algorithm=MD5, uri="sip:+87654321@test.local", nonce="2b5e06e094747daf2b5e06e0f11506e0b026768435f3086660000a7f39479e55e8e0bdbb", response="c73442a0ac4dcca88cca322b48d2019d" Date: Tue, 03 Apr 2012 16:51:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 262 v=0 o=session.com 621634435 621634436 IN IP4 10.62.150.68 s=pbx session c=IN IP4 10.62.150.68 t=0 0 m=audio 20252 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22181 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 To: From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 103 INVITE Content-Length: 0 <-------------> [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 2 [ 36]: To: [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Apr 3 18:51:29] VERBOSE[15615] chan_sip.c: --- (7 headers 0 lines) --- [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: *** SIP TIMER: Cancelling retransmission #22181 - INVITE (got response) [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '77a598877fb2fdad09dc26f039c8facc@test.local' Request 103: Found [Apr 3 18:51:29] DEBUG[15615] chan_sip.c: SIP response 100 to standard invite [Apr 3 18:51:31] DEBUG[15615] chan_sip.c: Auto destroying SIP dialog 'c444c12f08dc4dcbb0f1d87efbed86d3' [Apr 3 18:51:31] DEBUG[15615] chan_sip.c: Destroying SIP dialog c444c12f08dc4dcbb0f1d87efbed86d3 [Apr 3 18:51:31] VERBOSE[15615] chan_sip.c: Really destroying SIP dialog 'c444c12f08dc4dcbb0f1d87efbed86d3' Method: OPTIONS [Apr 3 18:51:33] WARNING[24574][C-00000002] res_srtp.c: SRTP unprotect failed with: authentication failure 10 [Apr 3 18:51:35] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 To: ;tag=35897fef From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: Supported: 100rel,replaces CSeq: 103 INVITE Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE Reason: TSSI;cause=180;reason=0;announced Content-Length: 0 <-------------> [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 2 [ 49]: To: ;tag=35897fef [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 5 [ 39]: Contact: [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 6 [ 26]: Supported: 100rel,replaces [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 8 [115]: Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 9 [ 41]: Reason: TSSI;cause=180;reason=0;announced [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Apr 3 18:51:35] VERBOSE[15615] chan_sip.c: --- (11 headers 0 lines) --- [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag 35897fef [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '77a598877fb2fdad09dc26f039c8facc@test.local' Request 103: Found [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: SIP response 180 to standard invite [Apr 3 18:51:35] DEBUG[15615] chan_sip.c: build_route: Contact hop: [Apr 3 18:51:35] VERBOSE[15615] chan_sip.c: list_route: hop: [Apr 3 18:51:35] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - tas [Apr 3 18:51:35] DEBUG[15607] chan_sip.c: Checking device state for peer tas [Apr 3 18:51:35] DEBUG[15607] devicestate.c: Changing state for SIP/tas - state 1 (Not in use) [Apr 3 18:51:35] DEBUG[15607] devicestate.c: device 'SIP/tas' state '1' [Apr 3 18:51:35] VERBOSE[24574][C-00000002] app_dial.c: -- SIP/tas-00000004 is ringing [Apr 3 18:51:35] DEBUG[24574][C-00000002] rtp_engine.c: Setting early bridge SDP of 'SIP/pbx-00000003' with that of 'SIP/tas-00000004' [Apr 3 18:51:35] VERBOSE[24574][C-00000002] chan_sip.c: <--- Transmitting (no NAT) to 10.62.150.23:62847 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda;received=10.62.150.23 From: "user";epid=89E6832516;tag=15165c8a26 To: ;tag=as235f078c Call-ID: 67c95c9a-199f-4864-8722-c69b12389c7c CSeq: 22303 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 3 18:51:35] DEBUG[24574][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 180' onto TLS socket destined for 10.62.150.23:62847 [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 To: ;tag=35897fef From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: Supported: 100rel,replaces CSeq: 103 INVITE Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE Reason: TSSI;cause=200;reason=0;announced Content-Type: application/sdp Content-Disposition: session Content-Length: 211 v=0 o=vision 2890844526 2890844526 IN IP4 217.7.75.80 s=VisionSession t=0 0 m=audio 9284 RTP/AVP 8 101 c=IN IP4 217.7.75.55 a=sendrecv a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK46d63744 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 2 [ 49]: To: ;tag=35897fef [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 5 [ 39]: Contact: [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 6 [ 26]: Supported: 100rel,replaces [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 8 [115]: Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 9 [ 41]: Reason: TSSI;cause=200;reason=0;announced [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 11 [ 28]: Content-Disposition: session [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 12 [ 19]: Content-Length: 211 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Header 13 [ 0]: [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 0 [ 3]: v=0 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 1 [ 49]: o=vision 2890844526 2890844526 IN IP4 217.7.75.80 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 2 [ 15]: s=VisionSession [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 3 [ 5]: t=0 0 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 4 [ 26]: m=audio 9284 RTP/AVP 8 101 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 5 [ 20]: c=IN IP4 217.7.75.55 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 6 [ 10]: a=sendrecv [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 7 [ 10]: a=ptime:20 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: --- (13 headers 10 lines) --- [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag 35897fef [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Acked pending invite 103 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Stopping retransmission on '77a598877fb2fdad09dc26f039c8facc@test.local' of Request 103: Match Found [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: SIP response 200 to standard invite [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing session-level SDP o=vision 2890844526 2890844526 IN IP4 217.7.75.80... UNSUPPORTED. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing session-level SDP s=VisionSession... UNSUPPORTED. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Found RTP audio format 8 [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Setting payload 8 based on m type on 0xb6bf3da8 [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Found RTP audio format 101 [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Setting payload 101 based on m type on 0xb6bf3da8 [Apr 3 18:51:38] DEBUG[15615] netsock2.c: Splitting '217.7.75.55' into... [Apr 3 18:51:38] DEBUG[15615] netsock2.c: ...host '217.7.75.55' and port ''. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 217.7.75.55... OK. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Found audio description format PCMA for ID 8 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Incorporating payload 8 on 0xb6bf3da8 [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Incorporating payload 101 on 0xb6bf3da8 [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 3 18:51:38] DEBUG[15615] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8cc7d9c' [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Peer audio RTP is at port 217.7.75.55:9284 [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Copying payload 8 from 0xb6bf3da8 to 0x8cc7f48 [Apr 3 18:51:38] DEBUG[15615] rtp_engine.c: Copying payload 101 from 0xb6bf3da8 to 0x8cc7f48 [Apr 3 18:51:38] DEBUG[15615] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x8cc7d9c' [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: We're settling with these formats: (alaw) [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: We have an owner, now see if we need to change this call [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Updating call counter for outgoing call [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: build_route: Contact hop: [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: list_route: hop: [Apr 3 18:51:38] DEBUG[15615] netsock2.c: Splitting '217.7.75.80:5060' into... [Apr 3 18:51:38] DEBUG[15615] netsock2.c: ...host '217.7.75.80' and port '5060'. [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Strict routing enforced for session 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: set_destination: Parsing for address/port to send to [Apr 3 18:51:38] DEBUG[15615] netsock2.c: Splitting '217.7.75.80:5060' into... [Apr 3 18:51:38] DEBUG[15615] netsock2.c: ...host '217.7.75.80' and port '5060'. [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: set_destination: set destination to 217.7.75.80:5060 [Apr 3 18:51:38] VERBOSE[15615] chan_sip.c: Transmitting (no NAT) to 217.7.75.80:5060: ACK sip:TASP001@217.7.75.80:5060 SIP/2.0 Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK12c1c431 Max-Forwards: 70 From: "user" ;tag=as1be54ed8 To: ;tag=35897fef Contact: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 103 ACK Content-Length: 0 --- [Apr 3 18:51:38] DEBUG[15615] chan_sip.c: Trying to put 'ACK sip:TAS' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:38] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - tas [Apr 3 18:51:38] DEBUG[15607] chan_sip.c: Checking device state for peer tas [Apr 3 18:51:38] DEBUG[15607] devicestate.c: Changing state for SIP/tas - state 1 (Not in use) [Apr 3 18:51:38] DEBUG[15607] devicestate.c: device 'SIP/tas' state '1' [Apr 3 18:51:38] VERBOSE[24574][C-00000002] app_dial.c: -- SIP/tas-00000004 answered SIP/pbx-00000003 [Apr 3 18:51:38] DEBUG[24574][C-00000002] rtp_engine.c: Setting early bridge SDP of 'SIP/pbx-00000003' with that of 'SIP/tas-00000004' [Apr 3 18:51:38] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - pbx [Apr 3 18:51:38] DEBUG[15607] chan_sip.c: Checking device state for peer pbx [Apr 3 18:51:38] DEBUG[15607] devicestate.c: Changing state for SIP/pbx - state 1 (Not in use) [Apr 3 18:51:38] DEBUG[15607] devicestate.c: device 'SIP/pbx' state '1' [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: SIP answering channel: SIP/pbx-00000003 [Apr 3 18:51:38] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: Setting framing from config on incoming call [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Apr 3 18:51:38] VERBOSE[24574][C-00000002] chan_sip.c: Audio is at 8282 [Apr 3 18:51:38] VERBOSE[24574][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP [Apr 3 18:51:38] VERBOSE[24574][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Apr 3 18:51:38] VERBOSE[24574][C-00000002] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.62.150.23:62847 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK8c113cda;received=10.62.150.23 From: "user";epid=89E6832516;tag=15165c8a26 To: ;tag=as235f078c Call-ID: 67c95c9a-199f-4864-8722-c69b12389c7c CSeq: 22303 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 346 v=0 o=session.com 346530056 346530057 IN IP4 10.62.150.68 s=pbx session c=IN IP4 10.62.150.68 t=0 0 m=audio 8282 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ldD7DqsCBntYJxxgX/2S4xvpScleKz7EYOZzXTXB <------------> [Apr 3 18:51:38] DEBUG[24574][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 10.62.150.23:62847 [Apr 3 18:51:38] DEBUG[24574][C-00000002] features.c: bridge answer set, chan answer set [Apr 3 18:51:38] DEBUG[24574][C-00000002] features.c: Removing dialed interfaces datastore on SIP/tas-00000004 since we're bridging [Apr 3 18:51:38] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:38] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 0 [ 62]: ACK sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 1 [ 90]: FROM: ;epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 2 [ 63]: TO: ;tag=as235f078c [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 3 [ 15]: CSEQ: 22303 ACK [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bKe682ffc7 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 7 [ 17]: CONTENT-LENGTH: 0 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 8 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 9 [ 0]: [Apr 3 18:51:38] VERBOSE[24573] chan_sip.c: <--- SIP read from TLS:10.62.150.23:62847 ---> ACK sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 FROM: ;epid=89E6832516;tag=15165c8a26 TO: ;tag=as235f078c CSEQ: 22303 ACK CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bKe682ffc7 CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer <-------------> [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 0 [ 62]: ACK sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 1 [ 90]: FROM: ;epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 2 [ 63]: TO: ;tag=as235f078c [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 3 [ 15]: CSEQ: 22303 ACK [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bKe682ffc7 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 7 [ 17]: CONTENT-LENGTH: 0 [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Header 8 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:38] VERBOSE[24573] chan_sip.c: --- (9 headers 0 lines) --- [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: = Looking for Call ID: 67c95c9a-199f-4864-8722-c69b12389c7c (Checking From) --From tag 15165c8a26 --To-tag as235f078c [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Apr 3 18:51:38] DEBUG[24573] chan_sip.c: Stopping retransmission on '67c95c9a-199f-4864-8722-c69b12389c7c' of Response 22303: Match Not Found [Apr 3 18:51:40] WARNING[24574][C-00000002] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Apr 3 18:51:42] WARNING[24574][C-00000002] res_srtp.c: SRTP unprotect failed with: authentication failure 110 [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> INVITE sip:+11223344@10.62.150.68:5060 SIP/2.0 Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e Max-Forwards: 70 To: "user" ;tag=as1be54ed8 From: ;tag=35897fef Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: CSeq: 685 INVITE Content-Type: application/sdp Content-Disposition: session Content-Length: 211 v=0 o=vision 2890844526 2890844526 IN IP4 217.7.75.80 s=VisionSession t=0 0 m=audio 9284 RTP/AVP 8 101 c=IN IP4 217.7.75.55 a=sendrecv a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 0 [ 51]: INVITE sip:+11223344@10.62.150.68:5060 SIP/2.0 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 3 [ 70]: To: "user" ;tag=as1be54ed8 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 4 [ 51]: From: ;tag=35897fef [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 5 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 6 [ 39]: Contact: [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 7 [ 16]: CSeq: 685 INVITE [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 9 [ 28]: Content-Disposition: session [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 10 [ 19]: Content-Length: 211 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 11 [ 0]: [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 0 [ 3]: v=0 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 1 [ 49]: o=vision 2890844526 2890844526 IN IP4 217.7.75.80 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 2 [ 15]: s=VisionSession [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 3 [ 5]: t=0 0 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 4 [ 26]: m=audio 9284 RTP/AVP 8 101 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 5 [ 20]: c=IN IP4 217.7.75.55 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 6 [ 10]: a=sendrecv [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 7 [ 10]: a=ptime:20 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: --- (11 headers 10 lines) --- [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking From) --From tag 35897fef --To-tag as1be54ed8 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Apr 3 18:51:44] DEBUG[15615] netsock2.c: Splitting '217.7.75.80:5060' into... [Apr 3 18:51:44] DEBUG[15615] netsock2.c: ...host '217.7.75.80' and port '5060'. [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: Sending to 217.7.75.80:5060 (no NAT) [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Initializing initreq for method INVITE - callid 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Call 77a598877fb2fdad09dc26f039c8facc@test.local responded to our reinvite without changing SDP version; ignoring SDP. [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Got a SIP re-invite for call 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: SIP/tas-00000004: This call is UP.... [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: <--- Transmitting (no NAT) to 217.7.75.80:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e;received=217.7.75.80 From: ;tag=35897fef To: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 685 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Setting framing from config on incoming call [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: ** Our prefcodec: (alaw) [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: Audio is at 20252 [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: Adding codec 100004 (alaw) to SDP [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: <--- Reliably Transmitting (no NAT) to 217.7.75.80:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e;received=217.7.75.80 From: ;tag=35897fef To: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 685 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=session.com 621634435 621634436 IN IP4 10.62.150.68 s=pbx session c=IN IP4 10.62.150.68 t=0 0 m=audio 20252 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22183 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:44] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:44] DEBUG[24574][C-00000002] channel.c: Got a FRAME_CONTROL (32) frame on channel SIP/tas-00000004 [Apr 3 18:51:44] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:44] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:44] DEBUG[24574][C-00000002] channel.c: Bridge stops bridging channels SIP/pbx-00000003 and SIP/tas-00000004 [Apr 3 18:51:44] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:44] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> ACK sip:+11223344@10.62.150.68:5060 SIP/2.0 Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e To: "user" ;tag=as1be54ed8 From: ;tag=35897fef Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: CSeq: 685 ACK Content-Length: 0 <-------------> [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 0 [ 48]: ACK sip:+11223344@10.62.150.68:5060 SIP/2.0 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 217.7.75.80:5060;branch=z9hG4bK778589a8b459934ff9147ffff9b46074.525a350e [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 2 [ 70]: To: "user" ;tag=as1be54ed8 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 3 [ 51]: From: ;tag=35897fef [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 5 [ 39]: Contact: [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 6 [ 13]: CSeq: 685 ACK [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Apr 3 18:51:44] VERBOSE[15615] chan_sip.c: --- (8 headers 0 lines) --- [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking From) --From tag 35897fef --To-tag as1be54ed8 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22183 [Apr 3 18:51:44] DEBUG[15615] chan_sip.c: Stopping retransmission on '77a598877fb2fdad09dc26f039c8facc@test.local' of Response 685: Match Found [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 0 [ 62]: BYE sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 1 [ 90]: FROM: ;epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 2 [ 63]: TO: ;tag=as235f078c [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 3 [ 15]: CSEQ: 22304 BYE [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK6aeac0f6 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 7 [ 55]: CONTACT: [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 9 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 10 [ 0]: [Apr 3 18:51:45] VERBOSE[24573] chan_sip.c: <--- SIP read from TLS:10.62.150.23:62847 ---> BYE sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 FROM: ;epid=89E6832516;tag=15165c8a26 TO: ;tag=as235f078c CSEQ: 22304 BYE CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK6aeac0f6 CONTACT: CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer <-------------> [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 0 [ 62]: BYE sip:+87654321@10.62.150.68:5067;transport=TLS SIP/2.0 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 1 [ 90]: FROM: ;epid=89E6832516;tag=15165c8a26 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 2 [ 63]: TO: ;tag=as235f078c [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 3 [ 15]: CSEQ: 22304 BYE [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 4 [ 45]: CALL-ID: 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 6 [ 58]: VIA: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK6aeac0f6 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 7 [ 55]: CONTACT: [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0 [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Header 9 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Apr 3 18:51:45] VERBOSE[24573] chan_sip.c: --- (10 headers 0 lines) --- [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: = Looking for Call ID: 67c95c9a-199f-4864-8722-c69b12389c7c (Checking From) --From tag 15165c8a26 --To-tag as235f078c [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Initializing initreq for method BYE - callid 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:45] DEBUG[24573] netsock2.c: Splitting '10.62.150.23:62847' into... [Apr 3 18:51:45] DEBUG[24573] netsock2.c: ...host '10.62.150.23' and port '62847'. [Apr 3 18:51:45] VERBOSE[24573] chan_sip.c: Sending to 10.62.150.23:62847 (no NAT) [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Setting SIP_ALREADYGONE on dialog 67c95c9a-199f-4864-8722-c69b12389c7c [Apr 3 18:51:45] DEBUG[24573] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8cb22f4' [Apr 3 18:51:45] VERBOSE[24573] chan_sip.c: Scheduling destruction of SIP dialog '67c95c9a-199f-4864-8722-c69b12389c7c' in 32000 ms (Method: BYE) [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Received bye, issuing owner hangup [Apr 3 18:51:45] VERBOSE[24573] chan_sip.c: <--- Transmitting (no NAT) to 10.62.150.23:62847 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.62.150.23:62847;branch=z9hG4bK6aeac0f6;received=10.62.150.23 From: ;epid=89E6832516;tag=15165c8a26 To: ;tag=as235f078c Call-ID: 67c95c9a-199f-4864-8722-c69b12389c7c CSeq: 22304 BYE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Apr 3 18:51:45] DEBUG[24573] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 10.62.150.23:62847 [Apr 3 18:51:45] DEBUG[24574][C-00000002] channel.c: Didn't get a frame from channel: SIP/pbx-00000003 [Apr 3 18:51:45] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Apr 3 18:51:45] DEBUG[24574][C-00000002] channel.c: Bridge stops bridging channels SIP/pbx-00000003 and SIP/tas-00000004 [Apr 3 18:51:45] DEBUG[24574][C-00000002] channel.c: Hanging up channel 'SIP/tas-00000004' [Apr 3 18:51:45] DEBUG[24574][C-00000002] chan_sip.c: Hanging up zombie call. Be scared. [Apr 3 18:51:45] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8cc7d9c' [Apr 3 18:51:45] VERBOSE[24574][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '77a598877fb2fdad09dc26f039c8facc@test.local' in 6400 ms (Method: ACK) [Apr 3 18:51:45] DEBUG[24574][C-00000002] chan_sip.c: Strict routing enforced for session 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:45] VERBOSE[24574][C-00000002] chan_sip.c: set_destination: Parsing for address/port to send to [Apr 3 18:51:45] DEBUG[24574][C-00000002] netsock2.c: Splitting '217.7.75.80:5060' into... [Apr 3 18:51:45] DEBUG[24574][C-00000002] netsock2.c: ...host '217.7.75.80' and port '5060'. [Apr 3 18:51:45] VERBOSE[24574][C-00000002] chan_sip.c: set_destination: set destination to 217.7.75.80:5060 [Apr 3 18:51:45] VERBOSE[24574][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 217.7.75.80:5060: BYE sip:TASP001@217.7.75.80:5060 SIP/2.0 Via: SIP/2.0/UDP 10.62.150.68:5060;branch=z9hG4bK1c147896 Max-Forwards: 70 From: "user" ;tag=as1be54ed8 To: ;tag=35897fef Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local CSeq: 104 BYE Authorization: Digest username="fmc", realm="test.local", algorithm=MD5, uri="sip:TASP001@217.7.75.80:5060", nonce="2b5e06e094747daf2b5e06e0f11506e0b026768435f3086660000a7f39479e55e8e0bdbb", response="5ca84388269d71eda6febe5b3409fb92" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 3 18:51:45] DEBUG[24574][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22186 [Apr 3 18:51:45] DEBUG[24574][C-00000002] chan_sip.c: Trying to put 'BYE sip:TAS' onto UDP socket destined for 217.7.75.80:5060 [Apr 3 18:51:45] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - tas [Apr 3 18:51:45] DEBUG[15607] chan_sip.c: Checking device state for peer tas [Apr 3 18:51:45] DEBUG[15607] devicestate.c: Changing state for SIP/tas - state 1 (Not in use) [Apr 3 18:51:45] DEBUG[15607] devicestate.c: device 'SIP/tas' state '1' [Apr 3 18:51:45] DEBUG[24574][C-00000002] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Apr 3 18:51:45] DEBUG[24574][C-00000002] pbx.c: Spawn extension (from-pbx,+87654321,2) exited non-zero on 'SIP/pbx-00000003' [Apr 3 18:51:45] VERBOSE[24574][C-00000002] pbx.c: == Spawn extension (from-pbx, +87654321, 2) exited non-zero on 'SIP/pbx-00000003' [Apr 3 18:51:45] DEBUG[24574][C-00000002] channel.c: Soft-Hanging up channel 'SIP/pbx-00000003' [Apr 3 18:51:45] DEBUG[24574][C-00000002] channel.c: Hanging up channel 'SIP/pbx-00000003' [Apr 3 18:51:45] DEBUG[24574][C-00000002] chan_sip.c: Hanging up zombie call. Be scared. [Apr 3 18:51:45] DEBUG[24574][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8cb22f4' [Apr 3 18:51:45] DEBUG[15607] devicestate.c: No provider found, checking channel drivers for SIP - pbx [Apr 3 18:51:45] DEBUG[15607] chan_sip.c: Checking device state for peer pbx [Apr 3 18:51:45] DEBUG[15607] devicestate.c: Changing state for SIP/pbx - state 1 (Not in use) [Apr 3 18:51:45] DEBUG[15607] devicestate.c: device 'SIP/pbx' state '1' [Apr 3 18:51:45] VERBOSE[15615] chan_sip.c: <--- SIP read from UDP:217.7.75.80:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c147896 To: ;tag=35897fef From: "user" ;tag=as1be54ed8 Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local Contact: CSeq: 104 BYE Content-Length: 0 <-------------> [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 10.62.150.68:5060;rport=5060;branch=z9hG4bK1c147896 [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 2 [ 49]: To: ;tag=35897fef [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 3 [ 72]: From: "user" ;tag=as1be54ed8 [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 4 [ 53]: Call-ID: 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 5 [ 39]: Contact: [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 6 [ 13]: CSeq: 104 BYE [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Apr 3 18:51:45] VERBOSE[15615] chan_sip.c: --- (8 headers 0 lines) --- [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: = Looking for Call ID: 77a598877fb2fdad09dc26f039c8facc@test.local (Checking To) --From tag as1be54ed8 --To-tag 35897fef [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22186 [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Stopping retransmission on '77a598877fb2fdad09dc26f039c8facc@test.local' of Request 104: Match Found [Apr 3 18:51:45] VERBOSE[15615] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Apr 3 18:51:45] DEBUG[15615] chan_sip.c: Destroying SIP dialog 77a598877fb2fdad09dc26f039c8facc@test.local [Apr 3 18:51:45] VERBOSE[15615] chan_sip.c: Really destroying SIP dialog '77a598877fb2fdad09dc26f039c8facc@test.local' Method: ACK [Apr 3 18:51:45] DEBUG[15615] rtp_engine.c: Destroyed RTP instance '0x8cc7d9c'