[2017-03-30 15:42:08.19955] VERBOSE[24876][C-000fecc4] netsock2.c: Using SIP RTP CoS mark 5 [2017-03-30 15:42:08.20024] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@incomming:1] NoOp("SIP/sw1_to_smg2016-001cedaf", "INCOMMING CALL FROM 89127600023 TO 970227") in new stack [2017-03-30 15:42:08.20036] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@incomming:2] Gosub("SIP/sw1_to_smg2016-001cedaf", "Modificators_mod_1_sub,970227,1") in new stack [2017-03-30 15:42:08.20045] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Modificators_mod_1_sub:1] Return("SIP/sw1_to_smg2016-001cedaf", "") in new stack [2017-03-30 15:42:08.20059] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@incomming:3] Gosub("SIP/sw1_to_smg2016-001cedaf", "Dial_in_SIX-check_sub,970227,1") in new stack [2017-03-30 15:42:08.20072] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:1] GotoIf("SIP/sw1_to_smg2016-001cedaf", "0?600:2") in new stack [2017-03-30 15:42:08.20076] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,2) [2017-03-30 15:42:08.20078] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:2] GotoIf("SIP/sw1_to_smg2016-001cedaf", "0?100:3") in new stack [2017-03-30 15:42:08.20081] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,3) [2017-03-30 15:42:08.20082] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:3] GotoIf("SIP/sw1_to_smg2016-001cedaf", "0?200:4") in new stack [2017-03-30 15:42:08.20084] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,4) [2017-03-30 15:42:08.20086] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:4] GotoIf("SIP/sw1_to_smg2016-001cedaf", "0?500:5") in new stack [2017-03-30 15:42:08.20088] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,5) [2017-03-30 15:42:08.20091] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:5] GotoIf("SIP/sw1_to_smg2016-001cedaf", "0?300:6") in new stack [2017-03-30 15:42:08.20094] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,6) [2017-03-30 15:42:08.20127] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:6] GotoIf("SIP/sw1_to_smg2016-001cedaf", "1?400:900") in new stack [2017-03-30 15:42:08.20130] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-check_sub,970227,400) [2017-03-30 15:42:08.20131] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-check_sub:400] Goto("SIP/sw1_to_smg2016-001cedaf", "Dial_in_SIX-call_direct,970227,1") in new stack [2017-03-30 15:42:08.20134] VERBOSE[23867][C-000fecc4] pbx_builtins.c: Goto (Dial_in_SIX-call_direct,970227,1) [2017-03-30 15:42:08.20143] VERBOSE[23867][C-000fecc4] pbx.c: Executing [970227@Dial_in_SIX-call_direct:1] Dial("SIP/sw1_to_smg2016-001cedaf", "SIP/34673&SIP/55909") in new stack [2017-03-30 15:42:08.20222] VERBOSE[23867][C-000fecc4] netsock2.c: Using SIP RTP CoS mark 5 [2017-03-30 15:42:08.20288] VERBOSE[23867][C-000fecc4] netsock2.c: Using SIP RTP CoS mark 5 [2017-03-30 15:42:08.20336] VERBOSE[23867][C-000fecc4] chan_sip.c: Audio is at 8176 [2017-03-30 15:42:08.20337] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec ulaw to SDP [2017-03-30 15:42:08.20338] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec alaw to SDP [2017-03-30 15:42:08.20338] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec gsm to SDP [2017-03-30 15:42:08.20339] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec g729 to SDP [2017-03-30 15:42:08.20339] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2017-03-30 15:42:08.20343] VERBOSE[23867][C-000fecc4] chan_sip.c: Reliably Transmitting (NAT) to 148.251.118.84:5060: INVITE sip:disp_pechka@192.168.99.3:5060 SIP/2.0 Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK04bc29b0;rport Max-Forwards: 70 From: "89127600023" ;tag=as08f2f3b1 To: Contact: Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 102 INVITE User-Agent: MarkTEL Date: Thu, 30 Mar 2017 11:42:08 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "89127600023" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 337 v=0 o=root 62426880 62426880 IN IP4 217.14.196.29 s=MarkTEL Server c=IN IP4 217.14.196.29 t=0 0 m=audio 8176 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- [2017-03-30 15:42:08.20410] VERBOSE[23867][C-000fecc4] app_dial.c: Called SIP/34673 [2017-03-30 15:42:08.20422] VERBOSE[23867][C-000fecc4] chan_sip.c: Audio is at 29636 [2017-03-30 15:42:08.20423] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec ulaw to SDP [2017-03-30 15:42:08.20423] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec alaw to SDP [2017-03-30 15:42:08.20424] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec gsm to SDP [2017-03-30 15:42:08.20424] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding codec g729 to SDP [2017-03-30 15:42:08.20424] VERBOSE[23867][C-000fecc4] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2017-03-30 15:42:08.20427] VERBOSE[23867][C-000fecc4] chan_sip.c: Reliably Transmitting (NAT) to 148.251.118.84:5060: INVITE sip:disp_pechka@192.168.99.3:5060 SIP/2.0 Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK262e969a;rport Max-Forwards: 70 From: "89127600023" ;tag=as528541b6 To: Contact: Call-ID: 392b024966b79235712dd97f42e281e8@217.14.196.29:5060 CSeq: 102 INVITE User-Agent: MarkTEL Date: Thu, 30 Mar 2017 11:42:08 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "89127600023" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 340 v=0 o=root 374034052 374034052 IN IP4 217.14.196.29 s=MarkTEL Server c=IN IP4 217.14.196.29 t=0 0 m=audio 29636 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- [2017-03-30 15:42:08.20493] VERBOSE[23867][C-000fecc4] app_dial.c: Called SIP/55909 [2017-03-30 15:42:08.20496] VERBOSE[23867][C-000fecc4] app_dial.c: SIP/55909-001cedb1 connected line has changed. Saving it until answer for SIP/sw1_to_smg2016-001cedaf [2017-03-30 15:42:08.20498] VERBOSE[23867][C-000fecc4] app_dial.c: SIP/34673-001cedb0 connected line has changed. Saving it until answer for SIP/sw1_to_smg2016-001cedaf [2017-03-30 15:42:08.26933] VERBOSE[24876] chan_sip.c: <--- SIP read from UDP:148.251.118.84:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK04bc29b0;received=217.14.196.29;rport=5060 From: "89127600023" ;tag=as08f2f3b1 To: Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.8.0~dfsg-0~ppa2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [2017-03-30 15:42:08.26936] VERBOSE[24876] chan_sip.c: --- (12 headers 0 lines) --- [2017-03-30 15:42:08.27127] VERBOSE[24876] chan_sip.c: <--- SIP read from UDP:148.251.118.84:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK262e969a;received=217.14.196.29;rport=5060 From: "89127600023" ;tag=as528541b6 To: Call-ID: 392b024966b79235712dd97f42e281e8@217.14.196.29:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.8.0~dfsg-0~ppa2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <-------------> [2017-03-30 15:42:08.27129] VERBOSE[24876] chan_sip.c: --- (12 headers 0 lines) --- [2017-03-30 15:42:08.27513] VERBOSE[24876] chan_sip.c: <--- SIP read from UDP:148.251.118.84:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK04bc29b0;received=217.14.196.29;rport=5060 From: "89127600023" ;tag=as08f2f3b1 To: ;tag=as16ba485c Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.8.0~dfsg-0~ppa2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 320 v=0 o=root 103021639 103021639 IN IP4 192.168.99.3 s=Asterisk PBX 13.8.0~dfsg-0~ppa2 c=IN IP4 192.168.99.3 t=0 0 m=audio 49916 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [2017-03-30 15:42:08.27516] VERBOSE[24876] chan_sip.c: --- (14 headers 14 lines) --- [2017-03-30 15:42:08.27523] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 0 [2017-03-30 15:42:08.27524] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 8 [2017-03-30 15:42:08.27524] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 18 [2017-03-30 15:42:08.27524] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 101 [2017-03-30 15:42:08.27525] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format PCMU for ID 0 [2017-03-30 15:42:08.27526] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format PCMA for ID 8 [2017-03-30 15:42:08.27526] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format G729 for ID 18 [2017-03-30 15:42:08.27527] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format telephone-event for ID 101 [2017-03-30 15:42:08.27529] VERBOSE[24876][C-000fecc4] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) [2017-03-30 15:42:08.27529] VERBOSE[24876][C-000fecc4] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2017-03-30 15:42:08.27533] VERBOSE[24876][C-000fecc4] chan_sip.c: Peer audio RTP is at port 192.168.99.3:49916 [2017-03-30 15:42:08.27537] VERBOSE[24876][C-000fecc4] sip/route.c: sip_route_dump: route/path hop: [2017-03-30 15:42:08.27541] VERBOSE[24876][C-000fecc4] chan_sip.c: Transmitting (NAT) to 148.251.118.84:5060: ACK sip:disp_pechka@192.168.99.3:5060 SIP/2.0 Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK65365fdf;rport Max-Forwards: 70 From: "89127600023" ;tag=as08f2f3b1 To: ;tag=as16ba485c Contact: Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 102 ACK User-Agent: MarkTEL Content-Length: 0 --- [2017-03-30 15:42:08.27546] VERBOSE[23867][C-000fecc4] app_dial.c: SIP/34673-001cedb0 connected line has changed. Saving it until answer for SIP/sw1_to_smg2016-001cedaf [2017-03-30 15:42:08.27551] VERBOSE[24876] chan_sip.c: <--- SIP read from UDP:148.251.118.84:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK262e969a;received=217.14.196.29;rport=5060 From: "89127600023" ;tag=as528541b6 To: ;tag=as42b495ad Call-ID: 392b024966b79235712dd97f42e281e8@217.14.196.29:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.8.0~dfsg-0~ppa2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 320 v=0 o=root 162867202 162867202 IN IP4 192.168.99.3 s=Asterisk PBX 13.8.0~dfsg-0~ppa2 c=IN IP4 192.168.99.3 t=0 0 m=audio 40008 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [2017-03-30 15:42:08.27553] VERBOSE[24876] chan_sip.c: --- (14 headers 14 lines) --- [2017-03-30 15:42:08.27556] VERBOSE[23867][C-000fecc4] app_dial.c: SIP/34673-001cedb0 answered SIP/sw1_to_smg2016-001cedaf [2017-03-30 15:42:08.27557] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 0 [2017-03-30 15:42:08.27557] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 8 [2017-03-30 15:42:08.27558] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 18 [2017-03-30 15:42:08.27558] VERBOSE[24876][C-000fecc4] chan_sip.c: Found RTP audio format 101 [2017-03-30 15:42:08.27558] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format PCMU for ID 0 [2017-03-30 15:42:08.27559] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format PCMA for ID 8 [2017-03-30 15:42:08.27559] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format G729 for ID 18 [2017-03-30 15:42:08.27560] VERBOSE[24876][C-000fecc4] chan_sip.c: Found audio description format telephone-event for ID 101 [2017-03-30 15:42:08.27561] VERBOSE[24876][C-000fecc4] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) [2017-03-30 15:42:08.27562] VERBOSE[24876][C-000fecc4] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2017-03-30 15:42:08.27564] VERBOSE[24876][C-000fecc4] chan_sip.c: Peer audio RTP is at port 192.168.99.3:40008 [2017-03-30 15:42:08.27567] VERBOSE[24876][C-000fecc4] sip/route.c: sip_route_dump: route/path hop: [2017-03-30 15:42:08.27569] VERBOSE[24876][C-000fecc4] chan_sip.c: Transmitting (NAT) to 148.251.118.84:5060: ACK sip:disp_pechka@192.168.99.3:5060 SIP/2.0 Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK57b501f6;rport Max-Forwards: 70 From: "89127600023" ;tag=as528541b6 To: ;tag=as42b495ad Contact: Call-ID: 392b024966b79235712dd97f42e281e8@217.14.196.29:5060 CSeq: 102 ACK User-Agent: MarkTEL Content-Length: 0 --- [2017-03-30 15:42:08.27580] VERBOSE[23867][C-000fecc4] chan_sip.c: Scheduling destruction of SIP dialog '392b024966b79235712dd97f42e281e8@217.14.196.29:5060' in 6400 ms (Method: INVITE) [2017-03-30 15:42:08.27733] VERBOSE[23868][C-000fecc4] bridge_channel.c: Channel SIP/34673-001cedb0 joined 'simple_bridge' basic-bridge [2017-03-30 15:42:08.27748] VERBOSE[23867][C-000fecc4] bridge_channel.c: Channel SIP/sw1_to_smg2016-001cedaf joined 'simple_bridge' basic-bridge [2017-03-30 15:42:08.32877] VERBOSE[23867][C-000fecc4] res_rtp_asterisk.c: 0x7f338ae6ec10 -- Probation passed - setting RTP source address to 217.14.196.30:25710 [2017-03-30 15:42:08.77938] VERBOSE[23868][C-000fecc4] res_rtp_asterisk.c: 0x7f334fa40350 -- Probation passed - setting RTP source address to 148.251.118.84:49916 [2017-03-30 15:42:08.79963] VERBOSE[23868][C-000fecc4] res_rtp_asterisk.c: 0x7f334fa40350 -- Probation passed - setting RTP source address to 148.251.118.84:49916 [2017-03-30 15:42:10.41088] VERBOSE[23867][C-000fecc4] bridge_channel.c: Channel SIP/sw1_to_smg2016-001cedaf left 'simple_bridge' basic-bridge [2017-03-30 15:42:10.41120] VERBOSE[23868][C-000fecc4] bridge_channel.c: Channel SIP/34673-001cedb0 left 'simple_bridge' basic-bridge [2017-03-30 15:42:10.41147] VERBOSE[23868][C-000fecc4] chan_sip.c: Scheduling destruction of SIP dialog '05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060' in 6400 ms (Method: INVITE) [2017-03-30 15:42:10.41152] VERBOSE[23867][C-000fecc4] pbx.c: Spawn extension (Dial_in_SIX-call_direct, 970227, 1) exited non-zero on 'SIP/sw1_to_smg2016-001cedaf' [2017-03-30 15:42:10.41186] VERBOSE[23868][C-000fecc4] chan_sip.c: Reliably Transmitting (NAT) to 148.251.118.84:5060: BYE sip:disp_pechka@192.168.99.3:5060 SIP/2.0 Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK6b22df15;rport Max-Forwards: 70 From: "89127600023" ;tag=as08f2f3b1 To: ;tag=as16ba485c Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 103 BYE User-Agent: MarkTEL X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2017-03-30 15:42:10.47628] VERBOSE[24876] chan_sip.c: <--- SIP read from UDP:148.251.118.84:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.196.29:5060;branch=z9hG4bK6b22df15;received=217.14.196.29;rport=5060 From: "89127600023" ;tag=as08f2f3b1 To: ;tag=as16ba485c Call-ID: 05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060 CSeq: 103 BYE Server: Asterisk PBX 13.8.0~dfsg-0~ppa2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [2017-03-30 15:42:10.47631] VERBOSE[24876] chan_sip.c: --- (10 headers 0 lines) --- [2017-03-30 15:42:10.47653] VERBOSE[24876] chan_sip.c: Really destroying SIP dialog '05c13bad44f9b9130aa7414e37cde942@217.14.196.29:5060' Method: INVITE