Asterisk 11.15.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 11.15.0 currently running on localhost (pid = 1756) localhost*CLI> Core debug is still 100. localhost*CLI>  == WebSocket connection from '10.156.2.55:12739' for protocol 'sip' accepted using version '13' localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> REGISTER sip:10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK9541296 Max-Forwards: 70 To: From: "8007" ;tag=2ggou1bbj5 Call-ID: 9v96glhfoi1vrhlkdkm1sf CSeq: 81 REGISTER Contact: ;reg-id=1;+sip.instance="";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK9541296;received=10.156.2.55 From: "8007" ;tag=2ggou1bbj5 To: ;tag=as6caa9ee1 Call-ID: 9v96glhfoi1vrhlkdkm1sf CSeq: 81 REGISTER Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0eee423a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '9v96glhfoi1vrhlkdkm1sf' in 32000 ms (Method: REGISTER) localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> REGISTER sip:10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK918844 Max-Forwards: 70 To: From: "8007" ;tag=2ggou1bbj5 Call-ID: 9v96glhfoi1vrhlkdkm1sf CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username="8007", realm="asterisk", nonce="0eee423a", uri="sip:10.156.2.56", response="947938aad1096c8097afc5afe812601c" Contact: ;reg-id=1;+sip.instance="";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 0 <-------------> --- (13 headers 0 lines) --- localhost*CLI>  == WebSocket connection from '10.156.2.55:12673' closed localhost*CLI>  -- Registered SIP '8007' at 10.156.2.55:12739 Reliably Transmitting (no NAT) to 10.156.2.55:12739: OPTIONS sip:2d92hv3q@192.0.2.35;transport=ws SIP/2.0 Via: SIP/2.0/WS 10.156.2.56:5060;branch=z9hG4bK5c81226c Max-Forwards: 70 From: "Unknown" ;tag=as15b4199e To: Contact: Call-ID: 36b740234916bf470412d5050dbe77dc@10.156.2.56:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:27:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK918844;received=10.156.2.55 From: "8007" ;tag=2ggou1bbj5 To: ;tag=as6caa9ee1 Call-ID: 9v96glhfoi1vrhlkdkm1sf CSeq: 82 REGISTER Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: ;expires=600 Date: Fri, 09 Jan 2015 08:27:21 GMT Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog '9v96glhfoi1vrhlkdkm1sf' in 32000 ms (Method: REGISTER) localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 10.156.2.56:5060;branch=z9hG4bK5c81226c To: ;tag=kpbf3dt950 From: "Unknown" ;tag=as15b4199e Call-ID: 36b740234916bf470412d5050dbe77dc@10.156.2.56:5060 CSeq: 102 OPTIONS Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Accept: application/sdp,application/dtmf-relay Supported: outbound Content-Length: 0 <-------------> localhost*CLI> --- (10 headers 0 lines) --- localhost*CLI> Really destroying SIP dialog '36b740234916bf470412d5050dbe77dc@10.156.2.56:5060' Method: OPTIONS localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> PUBLISH sip:8002@10.156.2.56 SIP/2.0 CSeq: 17 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK323f3d55-a80e-1910-9d7e-5cf9dd75cc57;rport User-Agent: Ekiga/4.0.2 From: Call-ID: 5f006a52-a80e-1910-9d77-5cf9dd75cc57@ss68 To: Expires: 300 Event: presence Content-Length: 484 Content-Type: application/pidf+xml Max-Forwards: 70 open sip:8002@10.156.2.56 I'm available using Ekiga 2015-01-09T16:19:37+08:00 <-------------> localhost*CLI> --- (12 headers 3 lines) --- localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> PUBLISH sip:8002@10.156.2.56 SIP/2.0 CSeq: 17 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK323f3d55-a80e-1910-9d7e-5cf9dd75cc57;rport User-Agent: Ekiga/4.0.2 From: Call-ID: 5f006a52-a80e-1910-9d77-5cf9dd75cc57@ss68 To: Expires: 300 Event: presence Content-Length: 484 Content-Type: application/pidf+xml Max-Forwards: 70 open sip:8002@10.156.2.56 I'm available using Ekiga 2015-01-09T16:19:37+08:00 <-------------> localhost*CLI> --- (12 headers 3 lines) --- localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> PUBLISH sip:8002@10.156.2.56 SIP/2.0 CSeq: 17 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK323f3d55-a80e-1910-9d7e-5cf9dd75cc57;rport User-Agent: Ekiga/4.0.2 From: Call-ID: 5f006a52-a80e-1910-9d77-5cf9dd75cc57@ss68 To: Expires: 300 Event: presence Content-Length: 484 Content-Type: application/pidf+xml Max-Forwards: 70 open sip:8002@10.156.2.56 I'm available using Ekiga 2015-01-09T16:19:37+08:00 <-------------> localhost*CLI> --- (12 headers 3 lines) --- localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> INVITE sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK5124570 Max-Forwards: 70 To: From: "8007" ;tag=ngjncjnqmt Call-ID: h08ha56dednb8osm6dlg CSeq: 2025 INVITE Contact: Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 1278 v=0 o=Mozilla-SIPUA-34.0.5 19294 0 IN IP4 192.0.2.14 s=SIP Call t=0 0 a=ice-ufrag:f9080812 a=ice-pwd:776b20a69cc6f0c678652800853778d4 a=fingerprint:sha-256 67:80:8C:84:70:8E:9F:58:25:AC:A5:3C:C1:2A:C3:93:10:6B:62:0A:E4:98:1D:C9:5E:07:D3:65:84:27:8D:D7 m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 192.0.2.14 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 10.156.2.55 54240 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 54241 typ host m=video 9 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 192.0.2.14 a=rtpmap:120 VP8/90000 a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42e01f;packetization-mode=1 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01f a=sendrecv a=rtcp-fb:120 nack a=rtcp-fb:120 nack pli a=rtcp-fb:120 ccm fir a=rtcp-fb:126 nack a=rtcp-fb:126 nack pli a=rtcp-fb:126 ccm fir a=rtcp-fb:97 nack a=rtcp-fb:97 nack pli a=rtcp-fb:97 ccm fir a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 10.156.2.55 54242 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 54243 typ host <-------------> localhost*CLI> --- (13 headers 43 lines) --- localhost*CLI> Using INVITE request as basis request - h08ha56dednb8osm6dlg localhost*CLI> Found peer '8007' for '8007' from 10.156.2.55:12739 localhost*CLI>  <--- Reliably Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK5124570;received=10.156.2.55 From: "8007" ;tag=ngjncjnqmt To: ;tag=as0512622a Call-ID: h08ha56dednb8osm6dlg CSeq: 2025 INVITE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c7b432" Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog 'h08ha56dednb8osm6dlg' in 6400 ms (Method: INVITE) localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> ACK sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK5124570 To: ;tag=as0512622a From: "8007" ;tag=ngjncjnqmt Call-ID: h08ha56dednb8osm6dlg CSeq: 2025 ACK <-------------> localhost*CLI> --- (6 headers 0 lines) --- localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> INVITE sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK3414795 Max-Forwards: 70 To: From: "8007" ;tag=ngjncjnqmt Call-ID: h08ha56dednb8osm6dlg CSeq: 2026 INVITE Authorization: Digest algorithm=MD5, username="8007", realm="asterisk", nonce="23c7b432", uri="sip:8002@10.156.2.56", response="fe7d896c7f1863903cba17afc1fc900c" Contact: Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 1278 v=0 o=Mozilla-SIPUA-34.0.5 19294 0 IN IP4 192.0.2.14 s=SIP Call t=0 0 a=ice-ufrag:f9080812 a=ice-pwd:776b20a69cc6f0c678652800853778d4 a=fingerprint:sha-256 67:80:8C:84:70:8E:9F:58:25:AC:A5:3C:C1:2A:C3:93:10:6B:62:0A:E4:98:1D:C9:5E:07:D3:65:84:27:8D:D7 m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 192.0.2.14 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 10.156.2.55 54240 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 54241 typ host m=video 9 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 192.0.2.14 a=rtpmap:120 VP8/90000 a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42e01f;packetization-mode=1 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01f a=sendrecv a=rtcp-fb:120 nack a=rtcp-fb:120 nack pli a=rtcp-fb:120 ccm fir a=rtcp-fb:126 nack a=rtcp-fb:126 nack pli a=rtcp-fb:126 ccm fir a=rtcp-fb:97 nack a=rtcp-fb:97 nack pli a=rtcp-fb:97 ccm fir a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 10.156.2.55 54242 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 54243 typ host <-------------> localhost*CLI> --- (14 headers 43 lines) --- localhost*CLI> Using INVITE request as basis request - h08ha56dednb8osm6dlg localhost*CLI> Found peer '8007' for '8007' from 10.156.2.55:12739 localhost*CLI>  == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Found RTP audio format 109 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found unknown media description format opus for ID 109 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Found RTP video format 120 Found RTP video format 126 Found RTP video format 97 Found video description format H264 for ID 126 Found video description format H264 for ID 97 Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.0.2.14:9 Peer video RTP is at port 192.0.2.14:9 Looking for 8002 in from-exten-sip (domain 10.156.2.56) list_route: hop: <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK3414795;received=10.156.2.55 From: "8007" ;tag=ngjncjnqmt To: Call-ID: h08ha56dednb8osm6dlg CSeq: 2026 INVITE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [8002@from-exten-sip:1] Verbose("SIP/8007-00000006", "10,'******Calling From '"8007" <8007>' To '8002 ") in new stack > '******Calling From '"8007" <8007>' To '8002 -- Executing [8002@from-exten-sip:2] Set("SIP/8007-00000006", "CHANNEL(language)=en") in new stack -- Executing [8002@from-exten-sip:3] Set("SIP/8007-00000006", "__FRI2_ROUTERLINE=1") in new stack -- Executing [8002@from-exten-sip:4] Set("SIP/8007-00000006", "__FRI2_EFET_ONETOUCHMONITOR=1") in new stack -- Executing [8002@from-exten-sip:5] AGI("SIP/8007-00000006", "agi://127.0.0.1/router?routerline=1") in new stack localhost*CLI>  -- AGI Script Executing Application: (ResetCDR) Options: () localhost*CLI>  -- AGI Script Executing Application: (AGI) Options: (agi://127.0.0.1/sysautomon?type=caller&number=8007) localhost*CLI>  -- AGI Script agi://127.0.0.1/sysautomon?type=caller&number=8007 completed, returning 0 localhost*CLI>  -- AGI Script agi://127.0.0.1/router?routerline=1 completed, returning 0 -- Executing [8002@from-exten-sip:6] ExecIf("SIP/8007-00000006", "0?Macro(failed-reason,NOVALID)") in new stack -- Executing [8002@from-exten-sip:7] GotoIf("SIP/8007-00000006", "1?app-internal,8002,1:app-outbound,8002,1") in new stack -- Goto (app-internal,8002,1) -- Executing [8002@app-internal:1] Verbose("SIP/8007-00000006", "10,'***********Internal************'") in new stack > '***********Internal************' -- Executing [8002@app-internal:2] Verbose("SIP/8007-00000006", "30,8002") in new stack > 8002 -- Executing [8002@app-internal:3] Verbose("SIP/8007-00000006", "30,") in new stack > -- Executing [8002@app-internal:4] Verbose("SIP/8007-00000006", "30,") in new stack > -- Executing [8002@app-internal:5] Verbose("SIP/8007-00000006", "30,") in new stack > -- Executing [8002@app-internal:6] ExecIf("SIP/8007-00000006", "0?Set(AGI_ROUTERPROCESS_EXTENSION=8002)") in new stack -- Executing [8002@app-internal:7] AGI("SIP/8007-00000006", "agi://127.0.0.1/dial_local?callednum=8002&typeof=&dialargs=") in new stack localhost*CLI>  -- AGI Script Executing Application: (AGI) Options: (agi://127.0.0.1/sysautomon?type=callee&number=8002) localhost*CLI>  -- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/10/SAM_1420792048921836_1420792049_573.WAV|ab|chmod 777 /var/spool/asterisk/monitor/10/SAM_1420792048921836_1420792049_573.WAV) localhost*CLI>  == Begin MixMonitor Recording SIP/8007-00000006 localhost*CLI>  -- AGI Script agi://127.0.0.1/sysautomon?type=callee&number=8002 completed, returning 0 localhost*CLI>  -- AGI Script agi://127.0.0.1/dial_local?callednum=8002&typeof=&dialargs= completed, returning 0 localhost*CLI>  -- Executing [8002@app-internal:8] Dial("SIP/8007-00000006", "sip/8002,30,tTKkM(answerhook)") in new stack localhost*CLI>  == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 localhost*CLI> Audio is at 16256 Video is at 10.156.2.56:18442 localhost*CLI> Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI> Reliably Transmitting (NAT) to 10.156.2.55:5060: INVITE sip:8002@10.156.2.55:5060 SIP/2.0 Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK56a02135;rport Max-Forwards: 70 From: "'8007'" ;tag=as0b99c15b To: Contact: Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 CSeq: 102 INVITE User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:27:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 366 v=0 o=root 1094715370 1094715370 IN IP4 10.156.2.56 s=Asterisk PBX 11.15.0 c=IN IP4 10.156.2.56 b=CT:2048 t=0 0 m=audio 16256 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 18442 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42E01F a=sendrecv --- -- Called sip/8002 localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK56a02135;rport=5060;received=10.156.2.56 From: "'8007'" ;tag=as0b99c15b Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 To: Content-Length: 0 <-------------> --- (7 headers 0 lines) --- localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> SIP/2.0 180 Ringing CSeq: 102 INVITE Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK56a02135;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "'8007'" ;tag=as0b99c15b Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 To: "8002" ;tag=93ea4155-a80e-1910-9d7e-5cf9dd75cc57 Contact: "8002" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- list_route: hop: -- SIP/8002-00000007 is ringing <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK3414795;received=10.156.2.55 From: "8007" ;tag=ngjncjnqmt To: ;tag=as6569b0ee Call-ID: h08ha56dednb8osm6dlg CSeq: 2026 INVITE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> PUBLISH sip:8002@10.156.2.56 SIP/2.0 CSeq: 17 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK323f3d55-a80e-1910-9d7e-5cf9dd75cc57;rport User-Agent: Ekiga/4.0.2 From: Call-ID: 5f006a52-a80e-1910-9d77-5cf9dd75cc57@ss68 To: Expires: 300 Event: presence Content-Length: 484 Content-Type: application/pidf+xml Max-Forwards: 70 open sip:8002@10.156.2.56 I'm available using Ekiga 2015-01-09T16:19:37+08:00 <-------------> localhost*CLI> --- (12 headers 3 lines) --- localhost*CLI>  <--- SIP read from UDP:10.156.2.253:5060 ---> OPTIONS sip:10.156.2.56 SIP/2.0 Via: SIP/2.0/UDP 10.156.2.253:5060;branch=z9hG4bK23eaa224 Max-Forwards: 70 From: "Unknown" ;tag=as6533b8c2 To: Contact: Call-ID: 694f8c79176a69a759f03e7e6133cece@10.156.2.253:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:26:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- localhost*CLI> Sending to 10.156.2.253:5060 (no NAT) Looking for s in from-trunk-other (domain 10.156.2.56) <--- Transmitting (no NAT) to 10.156.2.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.156.2.253:5060;branch=z9hG4bK23eaa224;received=10.156.2.253 From: "Unknown" ;tag=as6533b8c2 To: ;tag=as2780a55a Call-ID: 694f8c79176a69a759f03e7e6133cece@10.156.2.253:5060 CSeq: 102 OPTIONS Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '694f8c79176a69a759f03e7e6133cece@10.156.2.253:5060' in 32000 ms (Method: OPTIONS) localhost*CLI>  > 0xb7489378 -- Probation passed - setting RTP source address to 10.156.2.55:5070 localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> SIP/2.0 200 OK CSeq: 102 INVITE Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK56a02135;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "'8007'" ;tag=as0b99c15b Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 To: "8002" ;tag=93ea4155-a80e-1910-9d7e-5cf9dd75cc57 Contact: "8002" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Content-Length: 369 Content-Type: application/sdp v=0 o=- 1420792054 1 IN IP4 10.156.2.55 s=Ekiga/4.0.2 c=IN IP4 10.156.2.55 t=0 0 m=audio 5070 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:240 m=video 5072 RTP/AVP 99 b=AS:4096 b=TIAS:4096000 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 max-fs=3600;max-mbps=108000;profile-level-id=42801e <-------------> --- (11 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - (ulaw|alaw|h264), peer - audio=(alaw)/video=(h264)/text=(nothing), combined - (alaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.156.2.55:5070 Peer video RTP is at port 10.156.2.55:5072 localhost*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.156.2.55:5060 Transmitting (NAT) to 10.156.2.55:5060: ACK sip:8002@10.156.2.55:5060 SIP/2.0 Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK0985af74;rport Max-Forwards: 70 From: "'8007'" ;tag=as0b99c15b To: ;tag=93ea4155-a80e-1910-9d7e-5cf9dd75cc57 Contact: Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 CSeq: 102 ACK User-Agent: Asterisk Telephony Server Content-Length: 0 --- localhost*CLI>  -- SIP/8002-00000007 answered SIP/8007-00000006 localhost*CLI>  -- Executing [s@macro-answerhook:1] AGI("SIP/8002-00000007", "agi://127.0.0.1/answerhook") in new stack localhost*CLI>  > 0xb7489378 -- Probation passed - setting RTP source address to 10.156.2.55:5070 localhost*CLI>  -- AGI Script agi://127.0.0.1/answerhook completed, returning 0 localhost*CLI> Audio is at 13816 Video is at 10.156.2.56:14350 localhost*CLI> Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding video codec 200004 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI>  <--- Reliably Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK3414795;received=10.156.2.55 From: "8007" ;tag=ngjncjnqmt To: ;tag=as6569b0ee Call-ID: h08ha56dednb8osm6dlg CSeq: 2026 INVITE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 1136 v=0 o=root 2085420148 2085420148 IN IP4 10.156.2.56 s=Asterisk PBX 11.15.0 c=IN IP4 10.156.2.56 b=CT:2048 t=0 0 m=audio 13816 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:01be11e97c0b327c3f6a16ec2fea8ca2 a=ice-pwd:11af51503df8d1f14a1655c851312b37 a=candidate:Ha9c0238 1 UDP 2130706431 10.156.2.56 13816 typ host a=candidate:Ha9c0238 2 UDP 2130706430 10.156.2.56 13817 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 A1:7B:6E:5B:92:CE:9D:14:66:47:A0:A5:8C:B7:21:80:8E:E2:BF:47:B8:69:1C:67:E6:BF:0E:37:09:A4:73:44 a=sendrecv m=video 14350 UDP/TLS/RTP/SAVPF 97 a=ice-ufrag:4ecc217c778174de6e7208ed5bc591d5 a=ice-pwd:52a7b49b0b1ec05f6a35d41b7ef66386 a=candidate:Ha9c0238 1 UDP 2130706431 10.156.2.56 14350 typ host a=candidate:Ha9c0238 2 UDP 2130706430 10.156.2.56 14351 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 A1:7B:6E:5B:92:CE:9D:14:66:47:A0:A5:8C:B7:21:80:8E:E2:BF:47:B8:69:1C:67:E6:BF:0E:37:09:A4:73:44 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42E01F a=sendrecv <------------> localhost*CLI>  > 0xb74088c0 -- Probation passed - setting RTP source address to 10.156.2.55:54240 localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> ACK sip:8002@10.156.2.56:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.0.2.35;branch=z9hG4bK2583071 Max-Forwards: 70 To: ;tag=as6569b0ee From: "8007" ;tag=ngjncjnqmt Call-ID: h08ha56dednb8osm6dlg CSeq: 2026 ACK Supported: outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 0 <-------------> localhost*CLI> --- (10 headers 0 lines) --- localhost*CLI> Reliably Transmitting (NAT) to 10.156.2.55:5060: OPTIONS sip:8002@10.156.2.55:5060 SIP/2.0 Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK4e250df4;rport Max-Forwards: 70 From: "Unknown" ;tag=as6a025bed To: Contact: Call-ID: 016c5e821a35b76c674bce8e474afc7e@10.156.2.56:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:27:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK4e250df4;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "Unknown" ;tag=as6a025bed Call-ID: 016c5e821a35b76c674bce8e474afc7e@10.156.2.56:5060 To: Contact: Content-Length: 0 <-------------> localhost*CLI> --- (9 headers 0 lines) --- localhost*CLI> Really destroying SIP dialog '016c5e821a35b76c674bce8e474afc7e@10.156.2.56:5060' Method: OPTIONS localhost*CLI>  > 0xb7410ac8 -- Probation passed - setting RTP source address to 10.156.2.55:54242 localhost*CLI>  > 0xb7490590 -- Probation passed - setting RTP source address to 10.156.2.55:5072 localhost*CLI> Reliably Transmitting (no NAT) to 127.0.0.1:5060: OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d40bf6c Max-Forwards: 70 From: "Unknown" ;tag=as69e73683 To: Contact: Call-ID: 20de4298333788703a63f9490fdf4277@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:27:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:127.0.0.1:5060 ---> OPTIONS sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d40bf6c Max-Forwards: 70 From: "Unknown" ;tag=as69e73683 To: Contact: Call-ID: 20de4298333788703a63f9490fdf4277@127.0.0.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 08:27:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (13 headers 0 lines) --- localhost*CLI> Sending to 127.0.0.1:5060 (no NAT) Looking for s in from-trunk-other (domain 127.0.0.1) <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d40bf6c;received=127.0.0.1 From: "Unknown" ;tag=as69e73683 To: ;tag=as3f0f80ff Call-ID: 20de4298333788703a63f9490fdf4277@127.0.0.1:5060 CSeq: 102 OPTIONS Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '20de4298333788703a63f9490fdf4277@127.0.0.1:5060' in 32000 ms (Method: OPTIONS) <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d40bf6c;received=127.0.0.1 From: "Unknown" ;tag=as69e73683 To: ;tag=as3f0f80ff Call-ID: 20de4298333788703a63f9490fdf4277@127.0.0.1:5060 CSeq: 102 OPTIONS Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '20de4298333788703a63f9490fdf4277@127.0.0.1:5060' Method: OPTIONS localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> BYE sip:8007@10.156.2.56:5060 SIP/2.0 CSeq: 1 BYE Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK81ab5555-a80e-1910-9d7f-5cf9dd75cc57;rport User-Agent: Ekiga/4.0.2 From: ;tag=93ea4155-a80e-1910-9d7e-5cf9dd75cc57 Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 To: "'8007'" ;tag=as0b99c15b Contact: "8002" Content-Length: 0 Max-Forwards: 70 <-------------> --- (10 headers 0 lines) --- Sending to 10.156.2.55:5060 (NAT) localhost*CLI> Scheduling destruction of SIP dialog '70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060' in 6400 ms (Method: BYE) <--- Transmitting (NAT) to 10.156.2.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bK81ab5555-a80e-1910-9d7f-5cf9dd75cc57;received=10.156.2.55;rport=5060 From: ;tag=93ea4155-a80e-1910-9d7e-5cf9dd75cc57 To: "'8007'" ;tag=as0b99c15b Call-ID: 70f3a3267d75994e6ce7ece66443e7de@10.156.2.56:5060 CSeq: 1 BYE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> localhost*CLI>  -- Executing [h@app-internal:1] AGI("SIP/8007-00000006", "agi://127.0.0.1/dial_status") in new stack localhost*CLI>  -- AGI Script agi://127.0.0.1/dial_status completed, returning 0 -- Executing [h@app-internal:2] Hangup("SIP/8007-00000006", "") in new stack == Spawn extension (app-internal, h, 2) exited non-zero on 'SIP/8007-00000006' localhost*CLI>  == Spawn extension (app-internal, 8002, 8) exited non-zero on 'SIP/8007-00000006' localhost*CLI> Scheduling destruction of SIP dialog 'h08ha56dednb8osm6dlg' in 6400 ms (Method: INVITE) localhost*CLI> set_destination: Parsing for address/port to send to localhost*CLI> set_destination: URI is for WebSocket, we can't set destination localhost*CLI> Reliably Transmitting (no NAT) to 10.156.2.55:5060: BYE sip:2d92hv3q@192.0.2.35;transport=ws;ob SIP/2.0 Via: SIP/2.0/WS 10.156.2.56:5060;branch=z9hG4bK7ef6c862 Max-Forwards: 70 From: ;tag=as6569b0ee To: "8007" ;tag=ngjncjnqmt Call-ID: h08ha56dednb8osm6dlg CSeq: 102 BYE User-Agent: Asterisk Telephony Server Proxy-Authorization: Digest username="8007", realm="asterisk", algorithm=MD5, uri="sip:10.156.2.56", nonce="23c7b432", response="0c4426251a5ce95ef695d82ff6e67ac3" Reason: Q.850;cause=16 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- localhost*CLI>  == MixMonitor close filestream (mixed) localhost*CLI>  == Executing [chmod 777 /var/spool/asterisk/monitor/10/SAM_1420792048921836_1420792049_573.WAV] localhost*CLI>  == End MixMonitor Recording SIP/8007-00000006 localhost*CLI>  <--- SIP read from WS:10.156.2.55:12739 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 10.156.2.56:5060;branch=z9hG4bK7ef6c862 To: "8007" ;tag=ngjncjnqmt From: ;tag=as6569b0ee Call-ID: h08ha56dednb8osm6dlg CSeq: 102 BYE Supported: outbound Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> SIP Response message for INCOMING dialog BYE arrived localhost*CLI> Really destroying SIP dialog 'h08ha56dednb8osm6dlg' Method: INVITE localhost*CLI>