food*CLI> core set debug 9 Core debug was OFF and is now 9. food*CLI> sip set debug peer oper1 SIP Debugging Enabled for IP: 195.38.184.151 Really destroying SIP dialog '4839a4ab-cc05-bcc9-31d3-f2239c72fe27' Method: REGISTER Audio is at 15008 Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 195.38.184.151:32513: INVITE sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 212.42.122.200:5060;branch=z9hG4bK06b43863;rport Max-Forwards: 70 From: ;tag=as3b44a920 To: Contact: Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 INVITE User-Agent: AstPbx Date: Wed, 12 Feb 2014 05:18:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 391 v=0 o=root 2070050958 2070050958 IN IP4 212.42.122.200 s=Asterisk PBX SVN-branch-11-r407874 c=IN IP4 212.42.122.200 t=0 0 m=audio 15008 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+utjNKnXZ3jeFrNJy1lNo9HH2fHVblq5a9vW+F8n --- <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK06b43863 From: ;tag=as3b44a920 To: Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK06b43863 From: ;tag=as3b44a920 To: ;tag=8uBe6dbPMC2JEsn4Qyyr Contact: Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-------------> --- (9 headers 0 lines) --- list_route: hop: <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 603 Failed to get local SDP Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK06b43863 From: ;tag=as3b44a920 To: ;tag=8uBe6dbPMC2JEsn4Qyyr Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 Reason: SIP; cause=603; text="Failed to get local SDP" <-------------> --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: URI is for WebSocket, we can't set destination Transmitting (NAT) to 195.38.184.151:32513: ACK sip:oper1@df7jal23ls0d.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 212.42.122.200:5060;branch=z9hG4bK06b43863;rport Max-Forwards: 70 From: ;tag=as3b44a920 To: ;tag=8uBe6dbPMC2JEsn4Qyyr Contact: Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 ACK User-Agent: AstPbx Content-Length: 0 --- Really destroying SIP dialog '795949f01a3a189b00da88491a93823e@212.42.122.200:5060' Method: INVITE Audio is at 17406 Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 195.38.184.151:32513: INVITE sip:oper1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0 Via: SIP/2.0/WS 212.42.122.200:5060;branch=z9hG4bK7fa47621;rport Max-Forwards: 70 From: ;tag=as634129b4 To: Contact: Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 INVITE User-Agent: AstPbx Date: Wed, 12 Feb 2014 05:18:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 389 v=0 o=root 456270486 456270486 IN IP4 212.42.122.200 s=Asterisk PBX SVN-branch-11-r407874 c=IN IP4 212.42.122.200 t=0 0 m=audio 17406 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:tPTjyU9X7bRNelMklRuvnfbzKDm/SwaTelVqdwLP --- <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 481 Dialog/Transaction Does Not Exist Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK06b43863 From: ;tag=as3b44a920 To: ;tag=8uBe6dbPMC2JEsn4Qyyr Call-ID: 795949f01a3a189b00da88491a93823e@212.42.122.200:5060 CSeq: 102 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK7fa47621 From: ;tag=as634129b4 To: Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK7fa47621 From: ;tag=as634129b4 To: ;tag=q7sJoxxZfDFmpyvOP6TT Contact: Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-------------> --- (9 headers 0 lines) --- list_route: hop: <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 603 Failed to get local SDP Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK7fa47621 From: ;tag=as634129b4 To: ;tag=q7sJoxxZfDFmpyvOP6TT Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 INVITE Content-Length: 0 Reason: SIP; cause=603; text="Failed to get local SDP" <-------------> --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: URI is for WebSocket, we can't set destination Transmitting (NAT) to 195.38.184.151:32513: ACK sip:oper1@df7jal23ls0d.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 212.42.122.200:5060;branch=z9hG4bK7fa47621;rport Max-Forwards: 70 From: ;tag=as634129b4 To: ;tag=q7sJoxxZfDFmpyvOP6TT Contact: Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 ACK User-Agent: AstPbx Content-Length: 0 --- Really destroying SIP dialog '769c771413ff91f55ac162852507856f@212.42.122.200:5060' Method: INVITE <--- SIP read from WS:195.38.184.151:32513 ---> SIP/2.0 481 Dialog/Transaction Does Not Exist Via: SIP/2.0/WS 212.42.122.200:5060;rport=5060;branch=z9hG4bK7fa47621 From: ;tag=as634129b4 To: ;tag=q7sJoxxZfDFmpyvOP6TT Call-ID: 769c771413ff91f55ac162852507856f@212.42.122.200:5060 CSeq: 102 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) ---