[root@pbx null]# asterisk -r [Sep 8 07:48:53] Asterisk 13.10.0-rc1, Copyright (C) 1999 - 2014, Digium, Inc. and others. [Sep 8 07:48:53] Created by Mark Spencer [Sep 8 07:48:53] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Sep 8 07:48:53] This is free software, with components licensed under the GNU General Public [Sep 8 07:48:53] License version 2 and other licenses; you are welcome to redistribute it under [Sep 8 07:48:53] certain conditions. Type 'core show license' for details. [Sep 8 07:48:53] ========================================================================= [Sep 8 07:48:53] Connected to Asterisk 13.10.0-rc1 currently running on pbx (pid = 20034) Core debug is still 2. pbx*CLI> sip set debug peer vega pbx*CLI> rtp set debug ip 62.221.34.22 RTP Debugging Enabled for address: 62.221.34.22:0 [Sep 8 07:49:13] Reliably Transmitting (no NAT) to 62.221.34.22:5060: [Sep 8 07:49:13] OPTIONS sip:62.221.34.22 SIP/2.0 [Sep 8 07:49:13] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK629de55e [Sep 8 07:49:13] Max-Forwards: 70 [Sep 8 07:49:13] From: "asterisk" ;tag=as4c1b4cf2 [Sep 8 07:49:13] To: [Sep 8 07:49:13] Contact: [Sep 8 07:49:13] Call-ID: 1f81dbc5330fecd1242352b52dee09b1@212.65.93.74:5060 [Sep 8 07:49:13] CSeq: 102 OPTIONS [Sep 8 07:49:13] User-Agent: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:13] Date: Thu, 08 Sep 2016 07:49:13 GMT [Sep 8 07:49:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:13] Supported: replaces [Sep 8 07:49:13] Content-Length: 0 [Sep 8 07:49:13] [Sep 8 07:49:13] [Sep 8 07:49:13] --- [Sep 8 07:49:13] [Sep 8 07:49:13] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:13] SIP/2.0 200 OK [Sep 8 07:49:13] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK629de55e [Sep 8 07:49:13] From: "asterisk" ;tag=as4c1b4cf2 [Sep 8 07:49:13] To: ;tag=a6e0f48 [Sep 8 07:49:13] Call-ID: 1f81dbc5330fecd1242352b52dee09b1@212.65.93.74:5060 [Sep 8 07:49:13] CSeq: 102 OPTIONS [Sep 8 07:49:13] Server: VegaTelecom [Sep 8 07:49:13] Contact: [Sep 8 07:49:13] Content-Length: 0 [Sep 8 07:49:13] [Sep 8 07:49:13] <-------------> [Sep 8 07:49:13] --- (9 headers 0 lines) --- [Sep 8 07:49:13] Really destroying SIP dialog '1f81dbc5330fecd1242352b52dee09b1@212.65.93.74:5060' Method: OPTIONS [Sep 8 07:49:28] [Sep 8 07:49:28] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:28] INVITE sip:380445008353@212.65.93.74:5060;user=phone SIP/2.0 [Sep 8 07:49:28] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKcoaptu00e0e0qpshr4j0.1 [Sep 8 07:49:28] From: ;tag=1545581 [Sep 8 07:49:28] To: [Sep 8 07:49:28] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:28] CSeq: 1 INVITE [Sep 8 07:49:28] Max-Forwards: 68 [Sep 8 07:49:28] Supported: timer [Sep 8 07:49:28] P-Charging-Vector: icid-value=5D8EC560-0000-0000-0000-000057D11808 [Sep 8 07:49:28] Contact: [Sep 8 07:49:28] User-Agent: VegaTelecom [Sep 8 07:49:28] Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK [Sep 8 07:49:28] Content-Type: application/sdp [Sep 8 07:49:28] Content-Length: 484 [Sep 8 07:49:28] [Sep 8 07:49:28] v=0 [Sep 8 07:49:28] o=MG4000|2.0 191772 339560 IN IP4 62.221.34.22 [Sep 8 07:49:28] s=- [Sep 8 07:49:28] c=IN IP4 62.221.34.22 [Sep 8 07:49:28] t=0 0 [Sep 8 07:49:28] m=audio 32692 RTP/AVP 8 18 0 4 98 96 97 108 106 101 [Sep 8 07:49:28] a=rtpmap:98 G.729a/8000 [Sep 8 07:49:28] a=rtpmap:96 G.729ab/8000 [Sep 8 07:49:28] a=rtpmap:97 G.729b/8000 [Sep 8 07:49:28] a=rtpmap:108 G.723.1-L/8000 [Sep 8 07:49:28] a=rtpmap:106 G.723.1a-L/8000 [Sep 8 07:49:28] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:28] a=fmtp:101 0-15 [Sep 8 07:49:28] a=fmtp:4 annexa=no [Sep 8 07:49:28] a=fmtp:18 annexb=no [Sep 8 07:49:28] a=ptime:10 [Sep 8 07:49:28] a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0 [Sep 8 07:49:28] a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG [Sep 8 07:49:28] <-------------> [Sep 8 07:49:28] --- (14 headers 18 lines) --- [Sep 8 07:49:28] Sending to 62.221.34.22:5060 (NAT) [Sep 8 07:49:28] Sending to 62.221.34.22:5060 (NAT) [Sep 8 07:49:28] Using INVITE request as basis request - FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:28] Found peer 'vega' for '79152938288' from 62.221.34.22:5060 [Sep 8 07:49:28] == Using SIP RTP CoS mark 5 [Sep 8 07:49:28] Found RTP audio format 8 [Sep 8 07:49:28] Found RTP audio format 18 [Sep 8 07:49:28] Found RTP audio format 0 [Sep 8 07:49:28] Found RTP audio format 4 [Sep 8 07:49:28] Found RTP audio format 98 [Sep 8 07:49:28] Found RTP audio format 96 [Sep 8 07:49:28] Found RTP audio format 97 [Sep 8 07:49:28] Found RTP audio format 108 [Sep 8 07:49:28] Found RTP audio format 106 [Sep 8 07:49:28] Found RTP audio format 101 [Sep 8 07:49:28] Found unknown media description format G.729a for ID 98 [Sep 8 07:49:28] Found unknown media description format G.729ab for ID 96 [Sep 8 07:49:28] Found unknown media description format G.729b for ID 97 [Sep 8 07:49:28] Found unknown media description format G.723.1-L for ID 108 [Sep 8 07:49:28] Found unknown media description format G.723.1a-L for ID 106 [Sep 8 07:49:28] Found audio description format telephone-event for ID 101 [Sep 8 07:49:28] Capabilities: us - (alaw), peer - audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw) [Sep 8 07:49:28] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 8 07:49:28] Peer audio RTP is at port 62.221.34.22:32692 [Sep 8 07:49:28] Looking for 380445008353 in from-trunk (domain 212.65.93.74) [Sep 8 07:49:28] sip_route_dump: route/path hop: [Sep 8 07:49:28] [Sep 8 07:49:28] <--- Transmitting (no NAT) to 62.221.34.22:5060 ---> [Sep 8 07:49:28] SIP/2.0 100 Trying [Sep 8 07:49:28] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKcoaptu00e0e0qpshr4j0.1;received=62.221.34.22 [Sep 8 07:49:28] From: ;tag=1545581 [Sep 8 07:49:28] To: [Sep 8 07:49:28] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:28] CSeq: 1 INVITE [Sep 8 07:49:28] Server: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:28] Supported: replaces [Sep 8 07:49:28] Contact: [Sep 8 07:49:28] Content-Length: 0 [Sep 8 07:49:28] [Sep 8 07:49:28] [Sep 8 07:49:28] <------------> [Sep 8 07:49:28] Incoming call from 79152938288 to 380445008353 via vega [Sep 8 07:49:28] Concurent lines on vega is 1 [Sep 8 07:49:28] == Writing record file to /data/records/2016/09/08/20160908_07-49-28_79152938288-380445008353-1473320968.288611.ogg [Sep 8 07:49:28] == Begin MixMonitor Recording SIP/vega-0000966e [Sep 8 07:49:28] WARNING[13982][C-00002f01]: format_ogg_vorbis.c:377 ogg_vorbis_seek: Seeking is not supported on OGG/Vorbis streams in writing mode! [Sep 8 07:49:28] dial-out.agi,810380937718373,1: [1473320968.288611] User 79152938288 dialed 810380937718373. [Sep 8 07:49:29] == Manager 'dial-out' logged on from 127.0.0.1 [Sep 8 07:49:29] == dial-out.agi,810380937718373,1: [1473320968.288611] Choosed best route: vega,beeline,mts [Sep 8 07:49:29] == dial-out.agi,810380937718373,1: [1473320968.288611] Set callerid to 380445008354 for provider vega [Sep 8 07:49:29] dial-out.agi,810380937718373,1: [1473320968.288611] Attempt #1 to dial 380937718373 via vega from 380445008354 (79152938288). Number of concurrent calls are 1 of 30 [Sep 8 07:49:29] == Using SIP RTP CoS mark 5 [Sep 8 07:49:29] Audio is at 39172 [Sep 8 07:49:29] Adding codec alaw to SDP [Sep 8 07:49:29] Adding non-codec 0x1 (telephone-event) to SDP [Sep 8 07:49:29] Reliably Transmitting (no NAT) to 62.221.34.22:5060: [Sep 8 07:49:29] INVITE sip:380937718373@62.221.34.22 SIP/2.0 [Sep 8 07:49:29] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:29] Max-Forwards: 70 [Sep 8 07:49:29] From: ;tag=as389b7d1e [Sep 8 07:49:29] To: [Sep 8 07:49:29] Contact: [Sep 8 07:49:29] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:29] CSeq: 102 INVITE [Sep 8 07:49:29] User-Agent: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:29] Date: Thu, 08 Sep 2016 07:49:29 GMT [Sep 8 07:49:29] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:29] Supported: replaces [Sep 8 07:49:29] Remote-Party-ID: "380445008354" ;party=calling;privacy=off;screen=no [Sep 8 07:49:29] Content-Type: application/sdp [Sep 8 07:49:29] Content-Length: 244 [Sep 8 07:49:29] [Sep 8 07:49:29] v=0 [Sep 8 07:49:29] o=root 1312175227 1312175227 IN IP4 212.65.93.74 [Sep 8 07:49:29] s=Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:29] c=IN IP4 212.65.93.74 [Sep 8 07:49:29] t=0 0 [Sep 8 07:49:29] m=audio 39172 RTP/AVP 8 101 [Sep 8 07:49:29] a=rtpmap:8 PCMA/8000 [Sep 8 07:49:29] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:29] a=fmtp:101 0-16 [Sep 8 07:49:29] a=maxptime:150 [Sep 8 07:49:29] a=sendrecv [Sep 8 07:49:29] [Sep 8 07:49:29] --- [Sep 8 07:49:29] [Sep 8 07:49:29] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:29] SIP/2.0 100 Trying [Sep 8 07:49:29] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:29] From: ;tag=as389b7d1e [Sep 8 07:49:29] To: [Sep 8 07:49:29] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:29] CSeq: 102 INVITE [Sep 8 07:49:29] [Sep 8 07:49:29] <-------------> [Sep 8 07:49:29] --- (6 headers 0 lines) --- [Sep 8 07:49:31] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:31] SIP/2.0 183 Session Progress [Sep 8 07:49:31] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:31] From: ;tag=as389b7d1e [Sep 8 07:49:31] To: ;tag=96bc8b8 [Sep 8 07:49:31] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:31] CSeq: 102 INVITE [Sep 8 07:49:31] Server: VegaTelecom [Sep 8 07:49:31] Supported: timer,100rel [Sep 8 07:49:31] Contact: [Sep 8 07:49:31] Content-Length: 0 [Sep 8 07:49:31] [Sep 8 07:49:31] <-------------> [Sep 8 07:49:31] --- (10 headers 0 lines) --- [Sep 8 07:49:31] sip_route_dump: route/path hop: [Sep 8 07:49:31] [Sep 8 07:49:31] <--- Transmitting (no NAT) to 62.221.34.22:5060 ---> [Sep 8 07:49:31] SIP/2.0 180 Ringing [Sep 8 07:49:31] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKcoaptu00e0e0qpshr4j0.1;received=62.221.34.22 [Sep 8 07:49:31] From: ;tag=1545581 [Sep 8 07:49:31] To: ;tag=as265dad89 [Sep 8 07:49:31] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:31] CSeq: 1 INVITE [Sep 8 07:49:31] Server: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:31] Supported: replaces [Sep 8 07:49:31] Contact: [Sep 8 07:49:31] Content-Length: 0 [Sep 8 07:49:31] [Sep 8 07:49:31] [Sep 8 07:49:31] <------------> [Sep 8 07:49:31] [Sep 8 07:49:31] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:31] SIP/2.0 183 Session Progress [Sep 8 07:49:31] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:31] From: ;tag=as389b7d1e [Sep 8 07:49:31] To: ;tag=96bc8b8 [Sep 8 07:49:31] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:31] CSeq: 102 INVITE [Sep 8 07:49:31] Server: VegaTelecom [Sep 8 07:49:31] Supported: timer,100rel [Sep 8 07:49:31] Contact: [Sep 8 07:49:31] Content-Type: application/sdp [Sep 8 07:49:31] Content-Length: 284 [Sep 8 07:49:31] [Sep 8 07:49:31] v=0 [Sep 8 07:49:31] o=MG4000|2.0 313529 320707 IN IP4 62.221.34.22 [Sep 8 07:49:31] s=- [Sep 8 07:49:31] c=IN IP4 62.221.34.22 [Sep 8 07:49:31] t=0 0 [Sep 8 07:49:31] m=audio 41552 RTP/AVP 8 101 [Sep 8 07:49:31] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:31] a=fmtp:101 0-15 [Sep 8 07:49:31] a=ptime:20 [Sep 8 07:49:31] a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 [Sep 8 07:49:31] a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG [Sep 8 07:49:31] <-------------> [Sep 8 07:49:31] --- (11 headers 11 lines) --- [Sep 8 07:49:31] sip_route_dump: route/path hop: [Sep 8 07:49:31] Found RTP audio format 8 [Sep 8 07:49:31] Found RTP audio format 101 [Sep 8 07:49:31] Found audio description format telephone-event for ID 101 [Sep 8 07:49:31] Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Sep 8 07:49:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Sep 8 07:49:31] Peer audio RTP is at port 62.221.34.22:41552 [Sep 8 07:49:31] Audio is at 13130 [Sep 8 07:49:31] Adding codec alaw to SDP [Sep 8 07:49:31] Adding non-codec 0x1 (telephone-event) to SDP [Sep 8 07:49:31] [Sep 8 07:49:31] <--- Transmitting (no NAT) to 62.221.34.22:5060 ---> [Sep 8 07:49:31] SIP/2.0 183 Session Progress [Sep 8 07:49:31] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKcoaptu00e0e0qpshr4j0.1;received=62.221.34.22 [Sep 8 07:49:31] From: ;tag=1545581 [Sep 8 07:49:31] To: ;tag=as265dad89 [Sep 8 07:49:31] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:31] CSeq: 1 INVITE [Sep 8 07:49:31] Server: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:31] Supported: replaces [Sep 8 07:49:31] Contact: [Sep 8 07:49:31] Content-Type: application/sdp [Sep 8 07:49:31] Content-Length: 244 [Sep 8 07:49:31] [Sep 8 07:49:31] v=0 [Sep 8 07:49:31] o=root 1470959163 1470959163 IN IP4 212.65.93.74 [Sep 8 07:49:31] s=Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:31] c=IN IP4 212.65.93.74 [Sep 8 07:49:31] t=0 0 [Sep 8 07:49:31] m=audio 13130 RTP/AVP 8 101 [Sep 8 07:49:31] a=rtpmap:8 PCMA/8000 [Sep 8 07:49:31] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:31] a=fmtp:101 0-16 [Sep 8 07:49:31] a=maxptime:150 [Sep 8 07:49:31] a=sendrecv [Sep 8 07:49:31] [Sep 8 07:49:31] <------------> [Sep 8 07:49:32] [Sep 8 07:49:32] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:32] SIP/2.0 180 Ringing [Sep 8 07:49:32] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:32] From: ;tag=as389b7d1e [Sep 8 07:49:32] To: ;tag=96bc8b8 [Sep 8 07:49:32] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:32] CSeq: 102 INVITE [Sep 8 07:49:32] Server: VegaTelecom [Sep 8 07:49:32] Supported: timer,100rel [Sep 8 07:49:32] Contact: [Sep 8 07:49:32] Content-Type: application/sdp [Sep 8 07:49:32] Content-Length: 284 [Sep 8 07:49:32] [Sep 8 07:49:32] v=0 [Sep 8 07:49:32] o=MG4000|2.0 313529 320707 IN IP4 62.221.34.22 [Sep 8 07:49:32] s=- [Sep 8 07:49:32] c=IN IP4 62.221.34.22 [Sep 8 07:49:32] t=0 0 [Sep 8 07:49:32] m=audio 41552 RTP/AVP 8 101 [Sep 8 07:49:32] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:32] a=fmtp:101 0-15 [Sep 8 07:49:32] a=ptime:20 [Sep 8 07:49:32] a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 [Sep 8 07:49:32] a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG [Sep 8 07:49:32] <-------------> [Sep 8 07:49:32] --- (11 headers 11 lines) --- [Sep 8 07:49:32] sip_route_dump: route/path hop: [Sep 8 07:49:43] [Sep 8 07:49:43] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:43] SIP/2.0 200 OK [Sep 8 07:49:43] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK49ad14fd [Sep 8 07:49:43] From: ;tag=as389b7d1e [Sep 8 07:49:43] To: ;tag=96bc8b8 [Sep 8 07:49:43] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:43] CSeq: 102 INVITE [Sep 8 07:49:43] Server: VegaTelecom [Sep 8 07:49:43] Supported: timer,100rel [Sep 8 07:49:43] Contact: [Sep 8 07:49:43] Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK [Sep 8 07:49:43] Content-Type: application/sdp [Sep 8 07:49:43] Content-Length: 284 [Sep 8 07:49:43] [Sep 8 07:49:43] v=0 [Sep 8 07:49:43] o=MG4000|2.0 313529 320707 IN IP4 62.221.34.22 [Sep 8 07:49:43] s=- [Sep 8 07:49:43] c=IN IP4 62.221.34.22 [Sep 8 07:49:43] t=0 0 [Sep 8 07:49:43] m=audio 41552 RTP/AVP 8 101 [Sep 8 07:49:43] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:43] a=fmtp:101 0-15 [Sep 8 07:49:43] a=ptime:20 [Sep 8 07:49:43] a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 [Sep 8 07:49:43] a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG [Sep 8 07:49:43] <-------------> [Sep 8 07:49:43] --- (12 headers 11 lines) --- [Sep 8 07:49:43] sip_route_dump: route/path hop: [Sep 8 07:49:43] set_destination: Parsing for address/port to send to [Sep 8 07:49:43] set_destination: set destination to 62.221.34.22:5060 [Sep 8 07:49:43] Transmitting (no NAT) to 62.221.34.22:5060: [Sep 8 07:49:43] ACK sip:380937718373@62.221.34.22:5060;vtservice=b2buaservlet.siptosip;transport=udp SIP/2.0 [Sep 8 07:49:43] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK1c55e742 [Sep 8 07:49:43] Max-Forwards: 70 [Sep 8 07:49:43] From: ;tag=as389b7d1e [Sep 8 07:49:43] To: ;tag=96bc8b8 [Sep 8 07:49:43] Contact: [Sep 8 07:49:43] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:43] CSeq: 102 ACK [Sep 8 07:49:43] User-Agent: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:43] Content-Length: 0 [Sep 8 07:49:43] [Sep 8 07:49:43] [Sep 8 07:49:43] --- [Sep 8 07:49:43] Audio is at 13130 [Sep 8 07:49:43] Adding codec alaw to SDP [Sep 8 07:49:43] Adding non-codec 0x1 (telephone-event) to SDP [Sep 8 07:49:43] [Sep 8 07:49:43] <--- Reliably Transmitting (no NAT) to 62.221.34.22:5060 ---> [Sep 8 07:49:43] SIP/2.0 200 OK [Sep 8 07:49:43] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKcoaptu00e0e0qpshr4j0.1;received=62.221.34.22 [Sep 8 07:49:43] From: ;tag=1545581 [Sep 8 07:49:43] To: ;tag=as265dad89 [Sep 8 07:49:43] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:43] CSeq: 1 INVITE [Sep 8 07:49:43] Server: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:43] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:43] Supported: replaces [Sep 8 07:49:43] Contact: [Sep 8 07:49:43] Content-Type: application/sdp [Sep 8 07:49:43] Content-Length: 244 [Sep 8 07:49:43] [Sep 8 07:49:43] v=0 [Sep 8 07:49:43] o=root 1470959163 1470959163 IN IP4 212.65.93.74 [Sep 8 07:49:43] s=Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:43] c=IN IP4 212.65.93.74 [Sep 8 07:49:43] t=0 0 [Sep 8 07:49:43] m=audio 13130 RTP/AVP 8 101 [Sep 8 07:49:43] a=rtpmap:8 PCMA/8000 [Sep 8 07:49:43] a=rtpmap:101 telephone-event/8000 [Sep 8 07:49:43] a=fmtp:101 0-16 [Sep 8 07:49:43] a=maxptime:150 [Sep 8 07:49:43] a=sendrecv [Sep 8 07:49:43] [Sep 8 07:49:43] <------------> [Sep 8 07:49:43] [Sep 8 07:49:43] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:43] ACK sip:380445008353@212.65.93.74:5060 SIP/2.0 [Sep 8 07:49:43] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bK0k1e7820do10fo4v10q1.1 [Sep 8 07:49:43] From: ;tag=1545581 [Sep 8 07:49:43] To: ;tag=as265dad89 [Sep 8 07:49:43] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:43] CSeq: 1 ACK [Sep 8 07:49:43] User-Agent: VegaTelecom [Sep 8 07:49:43] Max-Forwards: 69 [Sep 8 07:49:43] Content-Length: 0 [Sep 8 07:49:43] [Sep 8 07:49:43] <-------------> [Sep 8 07:49:43] --- (9 headers 0 lines) --- [Sep 8 07:49:53] [Sep 8 07:49:53] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:53] BYE sip:380445008354@212.65.93.74:5060;PG=KIEVcAvtokonax01 SIP/2.0 [Sep 8 07:49:53] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKc8ih9o3048j0pmo62180.1 [Sep 8 07:49:53] From: ;tag=96bc8b8 [Sep 8 07:49:53] To: ;tag=as389b7d1e [Sep 8 07:49:53] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:53] CSeq: 2 BYE [Sep 8 07:49:53] Max-Forwards: 68 [Sep 8 07:49:53] Supported: timer,100rel [Sep 8 07:49:53] Reason: Q.850;cause=16;text="NormalRelease" [Sep 8 07:49:53] User-Agent: VegaTelecom [Sep 8 07:49:53] Content-Length: 0 [Sep 8 07:49:53] [Sep 8 07:49:53] <-------------> [Sep 8 07:49:53] --- (11 headers 0 lines) --- [Sep 8 07:49:53] Sending to 62.221.34.22:5060 (no NAT) [Sep 8 07:49:53] Scheduling destruction of SIP dialog '1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060' in 6400 ms (Method: BYE) [Sep 8 07:49:53] [Sep 8 07:49:53] <--- Transmitting (no NAT) to 62.221.34.22:5060 ---> [Sep 8 07:49:53] SIP/2.0 200 OK [Sep 8 07:49:53] Via: SIP/2.0/UDP 62.221.34.22:5060;branch=z9hG4bKc8ih9o3048j0pmo62180.1;received=62.221.34.22 [Sep 8 07:49:53] From: ;tag=96bc8b8 [Sep 8 07:49:53] To: ;tag=as389b7d1e [Sep 8 07:49:53] Call-ID: 1c2b3c5b49b575551c7822cd17ef453f@212.65.93.74:5060 [Sep 8 07:49:53] CSeq: 2 BYE [Sep 8 07:49:53] Server: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:53] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Sep 8 07:49:53] Supported: replaces [Sep 8 07:49:53] Content-Length: 0 [Sep 8 07:49:53] [Sep 8 07:49:53] [Sep 8 07:49:53] <------------> [Sep 8 07:49:53] WARNING[13981][C-00002f01]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Sep 8 07:49:53] WARNING[13981][C-00002f01]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Sep 8 07:49:53] dial-out.agi,810380937718373,1: [1473320968.288611] Hangup handler for 380937718373: dialed status is ANSWER: , hangupcause 16: , release B [Sep 8 07:49:53] == Manager 'dial-out' logged off from 127.0.0.1 [Sep 8 07:49:53] == Spawn extension (outgoing, 810380937718373, 1) exited non-zero on 'SIP/vega-0000966e' [Sep 8 07:49:53] Scheduling destruction of SIP dialog 'FCE99D91-85A3-46A5-8A34-53E266CD2100' in 6400 ms (Method: ACK) [Sep 8 07:49:53] == MixMonitor close filestream (mixed) [Sep 8 07:49:53] set_destination: Parsing for address/port to send to [Sep 8 07:49:53] set_destination: set destination to 62.221.34.22:5060 [Sep 8 07:49:53] Reliably Transmitting (no NAT) to 62.221.34.22:5060: [Sep 8 07:49:53] BYE sip:79152938288@62.221.34.22:5060;vtservice=b2buaservlet.siptosip;transport=udp SIP/2.0 [Sep 8 07:49:53] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK0ab5d4a5 [Sep 8 07:49:53] Max-Forwards: 70 [Sep 8 07:49:53] From: ;tag=as265dad89 [Sep 8 07:49:53] To: ;tag=1545581 [Sep 8 07:49:53] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:53] CSeq: 102 BYE [Sep 8 07:49:53] User-Agent: Asterisk PBX 13.10.0-rc1 [Sep 8 07:49:53] Reason: Q.850;cause=16 [Sep 8 07:49:53] X-Asterisk-HangupCause: Normal Clearing [Sep 8 07:49:53] X-Asterisk-HangupCauseCode: 16 [Sep 8 07:49:53] Content-Length: 0 [Sep 8 07:49:53] [Sep 8 07:49:53] [Sep 8 07:49:53] --- [Sep 8 07:49:53] == End MixMonitor Recording SIP/vega-0000966e [Sep 8 07:49:53] [Sep 8 07:49:53] <--- SIP read from UDP:62.221.34.22:5060 ---> [Sep 8 07:49:53] SIP/2.0 200 OK [Sep 8 07:49:53] Via: SIP/2.0/UDP 212.65.93.74:5060;branch=z9hG4bK0ab5d4a5 [Sep 8 07:49:53] From: ;tag=as265dad89 [Sep 8 07:49:53] To: ;tag=1545581 [Sep 8 07:49:53] Call-ID: FCE99D91-85A3-46A5-8A34-53E266CD2100 [Sep 8 07:49:53] CSeq: 102 BYE [Sep 8 07:49:53] Server: VegaTelecom [Sep 8 07:49:53] Contact: [Sep 8 07:49:53] Content-Length: 0 [Sep 8 07:49:53] [Sep 8 07:49:53] <-------------> [Sep 8 07:49:53] --- (9 headers 0 lines) --- [Sep 8 07:49:53] SIP Response message for INCOMING dialog BYE arrived [Sep 8 07:49:53] Really destroying SIP dialog 'FCE99D91-85A3-46A5-8A34-53E266CD2100' Method: ACK pbx*CLI> Disconnected from Asterisk server [Sep 8 07:50:02] Asterisk cleanly ending (0). [Sep 8 07:50:02] Executing last minute cleanups Asterisk ending (0). [root@pbx null]#