Asterisk 13.1.0, Copyright (C) 1999 - 2014, 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 13.1.0 currently running on localhost (pid = 1756) localhost*CLI> Core debug is still 100. localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> PUBLISH sip:8002@10.156.2.56 SIP/2.0 CSeq: 94 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bKf00a5565-a80e-1910-9d81-5cf9dd75cc57;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: 94 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bKf00a5565-a80e-1910-9d81-5cf9dd75cc57;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: 94 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bKf00a5565-a80e-1910-9d81-5cf9dd75cc57;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 <-------------> --- (12 headers 3 lines) --- localhost*CLI>  <--- SIP read from WS:10.156.2.55:13064 ---> INVITE sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK2343296 Max-Forwards: 70 To: From: "8007" ;tag=t7r0n6kctr Call-ID: 5ngq60shu77srpgvnisd CSeq: 3442 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: 1279 v=0 o=Mozilla-SIPUA-34.0.5 788 0 IN IP4 192.0.2.237 s=SIP Call t=0 0 a=ice-ufrag:b2522531 a=ice-pwd:719eebfaf0f00dc0a555334f27f946c0 a=fingerprint:sha-256 A3:E0:41:03:10:A9:B4:B1:A3:E9:46:71:83:EB:E4:E3:83:EF:2B:87:41:07:76:C5:FF:55:C2:79:88:49:AE:5D m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 192.0.2.237 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 50672 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 50673 typ host m=video 9 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 192.0.2.237 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 50674 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 50675 typ host <-------------> localhost*CLI> --- (13 headers 43 lines) --- Using INVITE request as basis request - 5ngq60shu77srpgvnisd Found peer '8007' for '8007' from 10.156.2.55:13064 <--- Reliably Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK2343296;received=10.156.2.55 From: "8007" ;tag=t7r0n6kctr To: ;tag=as2b2a4dbd Call-ID: 5ngq60shu77srpgvnisd CSeq: 3442 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="73206c92" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5ngq60shu77srpgvnisd' in 6400 ms (Method: INVITE) localhost*CLI>  <--- SIP read from WS:10.156.2.55:13064 ---> ACK sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK2343296 To: ;tag=as2b2a4dbd From: "8007" ;tag=t7r0n6kctr Call-ID: 5ngq60shu77srpgvnisd CSeq: 3442 ACK <-------------> --- (6 headers 0 lines) --- localhost*CLI>  <--- SIP read from WS:10.156.2.55:13064 ---> INVITE sip:8002@10.156.2.56 SIP/2.0 Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK9190082 Max-Forwards: 70 To: From: "8007" ;tag=t7r0n6kctr Call-ID: 5ngq60shu77srpgvnisd CSeq: 3443 INVITE Authorization: Digest algorithm=MD5, username="8007", realm="asterisk", nonce="73206c92", uri="sip:8002@10.156.2.56", response="94285fbe84b587e7b70a0c7a73ab4cdb" 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: 1279 v=0 o=Mozilla-SIPUA-34.0.5 788 0 IN IP4 192.0.2.237 s=SIP Call t=0 0 a=ice-ufrag:b2522531 a=ice-pwd:719eebfaf0f00dc0a555334f27f946c0 a=fingerprint:sha-256 A3:E0:41:03:10:A9:B4:B1:A3:E9:46:71:83:EB:E4:E3:83:EF:2B:87:41:07:76:C5:FF:55:C2:79:88:49:AE:5D m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 192.0.2.237 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 50672 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 50673 typ host m=video 9 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 192.0.2.237 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 50674 typ host a=candidate:0 2 UDP 2128609534 10.156.2.55 50675 typ host <-------------> --- (14 headers 43 lines) --- Using INVITE request as basis request - 5ngq60shu77srpgvnisd Found peer '8007' for '8007' from 10.156.2.55:13064 localhost*CLI>  == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 localhost*CLI>  == 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 audio 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 VP8 for ID 120 Found video description format H264 for ID 126 Found video description format H264 for ID 97 Capabilities: us - (alaw|ulaw|h264|vp8), peer - audio=(ulaw|alaw|g722|opus)/video=(h264|vp8)/text=(nothing), combined - (alaw|ulaw|h264|vp8) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.0.2.237:9 Peer video RTP is at port 192.0.2.237:9 localhost*CLI> Looking for 8002 in from-exten-sip (domain 10.156.2.56) localhost*CLI> sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK9190082;received=10.156.2.55 From: "8007" ;tag=t7r0n6kctr To: Call-ID: 5ngq60shu77srpgvnisd CSeq: 3443 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>  -- Executing [8002@from-exten-sip:1] Verbose("SIP/8007-00000002", "10,'******Calling From '"8007" <8007>' To '8002 ") in new stack localhost*CLI>  > '******Calling From '"8007" <8007>' To '8002 localhost*CLI>  -- Executing [8002@from-exten-sip:2] Set("SIP/8007-00000002", "CHANNEL(language)=en") in new stack localhost*CLI>  -- Executing [8002@from-exten-sip:3] Set("SIP/8007-00000002", "__FRI2_ROUTERLINE=1") in new stack localhost*CLI>  -- Executing [8002@from-exten-sip:4] Set("SIP/8007-00000002", "__FRI2_EFET_ONETOUCHMONITOR=1") in new stack localhost*CLI>  -- Executing [8002@from-exten-sip:5] AGI("SIP/8007-00000002", "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/8007-00000002", "0?Macro(failed-reason,NOVALID)") in new stack -- Executing [8002@from-exten-sip:7] GotoIf("SIP/8007-00000002", "1?app-internal,8002,1:app-outbound,8002,1") in new stack -- Goto (app-internal,8002,1) -- Executing [8002@app-internal:1] Verbose("SIP/8007-00000002", "10,'***********Internal************'") in new stack > '***********Internal************' -- Executing [8002@app-internal:2] Verbose("SIP/8007-00000002", "30,8002") in new stack > 8002 -- Executing [8002@app-internal:3] Verbose("SIP/8007-00000002", "30,") in new stack > -- Executing [8002@app-internal:4] Verbose("SIP/8007-00000002", "30,") in new stack > -- Executing [8002@app-internal:5] Verbose("SIP/8007-00000002", "30,") in new stack > -- Executing [8002@app-internal:6] ExecIf("SIP/8007-00000002", "0?Set(AGI_ROUTERPROCESS_EXTENSION=8002)") in new stack -- Executing [8002@app-internal:7] AGI("SIP/8007-00000002", "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_1420794748999165_1420794749_739.WAV|ab|chmod 777 /var/spool/asterisk/monitor/10/SAM_1420794748999165_1420794749_739.WAV) localhost*CLI>  == Begin MixMonitor Recording SIP/8007-00000002 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 -- Executing [8002@app-internal:8] Dial("SIP/8007-00000002", "sip/8002,30,tTKkM(answerhook)") in new stack == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 localhost*CLI> Audio is at 15888 Video is at 10.156.2.56:19386 Adding codec alaw to SDP Adding video codec h264 to SDP Adding video codec vp8 to SDP Adding codec ulaw to SDP Adding non-codec 0x1 (telephone-event) to SDP 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=z9hG4bK03724709;rport Max-Forwards: 70 From: "'8007'" ;tag=as48231c84 To: Contact: Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 CSeq: 102 INVITE User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 09:12:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 418 v=0 o=root 1853088542 1853088542 IN IP4 10.156.2.56 s=Asterisk PBX 13.1.0 c=IN IP4 10.156.2.56 b=CT:2048 t=0 0 m=audio 15888 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=maxptime:150 a=sendrecv m=video 19386 RTP/AVP 99 100 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42E01F a=rtpmap:100 VP8/90000 a=rtcp-fb:* ccm fir 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=z9hG4bK03724709;rport=5060;received=10.156.2.56 From: "'8007'" ;tag=as48231c84 Call-ID: 1c97300f3f62456c597168df204b0008@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=z9hG4bK03724709;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "'8007'" ;tag=as48231c84 Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 To: "8002" ;tag=fbfe5965-a80e-1910-9d81-5cf9dd75cc57 Contact: "8002" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sip_route_dump: route/path hop: localhost*CLI>  -- SIP/8002-00000003 is ringing <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK9190082;received=10.156.2.55 From: "8007" ;tag=t7r0n6kctr To: ;tag=as68e57b06 Call-ID: 5ngq60shu77srpgvnisd CSeq: 3443 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: 94 PUBLISH Via: SIP/2.0/UDP 10.156.2.55:5060;branch=z9hG4bKf00a5565-a80e-1910-9d81-5cf9dd75cc57;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>  > 0xb7408dd8 -- Probation passed - setting RTP source address to 10.156.2.55:5078 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=z9hG4bK03724709;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "'8007'" ;tag=as48231c84 Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 To: "8002" ;tag=fbfe5965-a80e-1910-9d81-5cf9dd75cc57 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=- 1420794754 1 IN IP4 10.156.2.55 s=Ekiga/4.0.2 c=IN IP4 10.156.2.55 t=0 0 m=audio 5078 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 5080 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 localhost*CLI> Found video description format H264 for ID 99 Capabilities: us - (alaw|ulaw|h264|vp8), 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 10.156.2.55:5078 Peer video RTP is at port 10.156.2.55:5080 sip_route_dump: route/path hop: 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=z9hG4bK0001137f;rport Max-Forwards: 70 From: "'8007'" ;tag=as48231c84 To: ;tag=fbfe5965-a80e-1910-9d81-5cf9dd75cc57 Contact: Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 CSeq: 102 ACK User-Agent: Asterisk Telephony Server Content-Length: 0 --- -- SIP/8002-00000003 answered SIP/8007-00000002 -- Executing [s@macro-answerhook:1] AGI("SIP/8002-00000003", "agi://127.0.0.1/answerhook") in new stack localhost*CLI>  > 0xb7408dd8 -- Probation passed - setting RTP source address to 10.156.2.55:5078 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=z9hG4bK02424990 Max-Forwards: 70 From: "Unknown" ;tag=as3b6f5d84 To: Contact: Call-ID: 1e7e357466e00e2f43be61d72ed87006@10.156.2.253:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 09:11:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> localhost*CLI> --- (13 headers 0 lines) --- 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=z9hG4bK02424990;received=10.156.2.253 From: "Unknown" ;tag=as3b6f5d84 To: ;tag=as438daa67 Call-ID: 1e7e357466e00e2f43be61d72ed87006@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 '1e7e357466e00e2f43be61d72ed87006@10.156.2.253:5060' in 32000 ms (Method: OPTIONS) localhost*CLI>  -- AGI Script agi://127.0.0.1/answerhook completed, returning 0 localhost*CLI> Audio is at 15836 Video is at 10.156.2.56:14308 Adding codec alaw to SDP Adding codec ulaw to SDP Adding video codec h264 to SDP Adding video codec vp8 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 192.0.2.137;branch=z9hG4bK9190082;received=10.156.2.55 From: "8007" ;tag=t7r0n6kctr To: ;tag=as68e57b06 Call-ID: 5ngq60shu77srpgvnisd CSeq: 3443 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: 742 v=0 o=root 920971968 920971968 IN IP4 10.156.2.56 s=Asterisk PBX 13.1.0 c=IN IP4 10.156.2.56 b=CT:2048 t=0 0 m=audio 15836 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=maxptime:150 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 14308 UDP/TLS/RTP/SAVPF 97 120 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=rtpmap:120 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> -- Channel SIP/8007-00000002 joined 'simple_bridge' basic-bridge localhost*CLI>  -- Channel SIP/8002-00000003 joined 'simple_bridge' basic-bridge localhost*CLI>  <--- SIP read from WS:10.156.2.55:13064 ---> ACK sip:8002@10.156.2.56:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK9160625 Max-Forwards: 70 To: ;tag=as68e57b06 From: "8007" ;tag=t7r0n6kctr Call-ID: 5ngq60shu77srpgvnisd CSeq: 3443 ACK Supported: outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 0 <-------------> --- (10 headers 0 lines) --- localhost*CLI>  <--- SIP read from WS:10.156.2.55:13064 ---> BYE sip:8002@10.156.2.56:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK3090475 Max-Forwards: 70 To: ;tag=as68e57b06 From: "8007" ;tag=t7r0n6kctr Call-ID: 5ngq60shu77srpgvnisd CSeq: 3444 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: outbound User-Agent: SIP.js/0.6.4-BB Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '5ngq60shu77srpgvnisd' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 10.156.2.55:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.0.2.137;branch=z9hG4bK3090475;received=10.156.2.55 From: "8007" ;tag=t7r0n6kctr To: ;tag=as68e57b06 Call-ID: 5ngq60shu77srpgvnisd CSeq: 3444 BYE Server: Asterisk Telephony Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/8007-00000002 left 'simple_bridge' basic-bridge == Spawn extension (app-internal, 8002, 8) exited non-zero on 'SIP/8007-00000002' -- Executing [h@app-internal:1] AGI("SIP/8007-00000002", "agi://127.0.0.1/dial_status") in new stack -- Channel SIP/8002-00000003 left 'simple_bridge' basic-bridge Scheduling destruction of SIP dialog '1c97300f3f62456c597168df204b0008@10.156.2.56:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.156.2.55:5060: BYE sip:8002@10.156.2.55:5060 SIP/2.0 Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK38ed8a19;rport Max-Forwards: 70 From: "'8007'" ;tag=as48231c84 To: ;tag=fbfe5965-a80e-1910-9d81-5cf9dd75cc57 Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 CSeq: 103 BYE User-Agent: Asterisk Telephony Server Reason: Q.850;cause=16 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- localhost*CLI>  <--- SIP read from UDP:10.156.2.55:5060 ---> SIP/2.0 200 OK CSeq: 103 BYE Via: SIP/2.0/UDP 10.156.2.56:5060;branch=z9hG4bK38ed8a19;rport From: "'8007'" ;tag=as48231c84 Call-ID: 1c97300f3f62456c597168df204b0008@10.156.2.56:5060 To: ;tag=fbfe5965-a80e-1910-9d81-5cf9dd75cc57 Content-Length: 0 <-------------> localhost*CLI> --- (7 headers 0 lines) --- Really destroying SIP dialog '1c97300f3f62456c597168df204b0008@10.156.2.56:5060' Method: INVITE localhost*CLI>  -- AGI Script agi://127.0.0.1/dial_status completed, returning 0 localhost*CLI>  -- Executing [h@app-internal:2] Hangup("SIP/8007-00000002", "") in new stack == Spawn extension (app-internal, h, 2) exited non-zero on 'SIP/8007-00000002' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/8007-00000002 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=z9hG4bK6f4a70f2;rport Max-Forwards: 70 From: "Unknown" ;tag=as752a643c To: Contact: Call-ID: 7496b9497f15f83733767d6a7e847625@10.156.2.56:5060 CSeq: 102 OPTIONS User-Agent: Asterisk Telephony Server Date: Fri, 09 Jan 2015 09:12:31 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=z9hG4bK6f4a70f2;rport=5060;received=10.156.2.56 User-Agent: Ekiga/4.0.2 From: "Unknown" ;tag=as752a643c Call-ID: 7496b9497f15f83733767d6a7e847625@10.156.2.56:5060 To: Contact: Content-Length: 0 <-------------> localhost*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '7496b9497f15f83733767d6a7e847625@10.156.2.56:5060' Method: OPTIONS localhost*CLI> Really destroying SIP dialog '27c2892a11ce776635d903847e06baf3@127.0.0.1:5060' Method: OPTIONS localhost*CLI> Really destroying SIP dialog '5ngq60shu77srpgvnisd' Method: BYE localhost*CLI>