pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Executing [3011@test#dial:1] Macro("SIP/test.3012-b7ebabe0", "stdexten") in new stack -- Executing [s@macro-stdexten:1] Set("SIP/test.3012-b7ebabe0", "ARRAY(client_id,type)=471,conc") in new stack -- Executing [s@macro-stdexten:2] Set("SIP/test.3012-b7ebabe0", "ARRAY(count,devices,ringt)=1,SIP/test.3011,20") in new stack -- Executing [s@macro-stdexten:3] GotoIf("SIP/test.3012-b7ebabe0", "1?one:many") in new stack -- Goto (macro-stdexten,s,15) -- Executing [s@macro-stdexten:15] Macro("SIP/test.3012-b7ebabe0", "dialdevice,SIP/test.3011,20") in new stack -- Executing [s@macro-dialdevice:1] Set("SIP/test.3012-b7ebabe0", "device=test.3011") in new stack -- Executing [s@macro-dialdevice:2] Set("SIP/test.3012-b7ebabe0", "devserver=pbx1") in new stack -- Executing [s@macro-dialdevice:3] Verbose("SIP/test.3012-b7ebabe0", "4,Dev Server: pbx1") in new stack > Dev Server: pbx1 -- Executing [s@macro-dialdevice:4] Verbose("SIP/test.3012-b7ebabe0", "4,My Server: pbx1") in new stack > My Server: pbx1 -- Executing [s@macro-dialdevice:5] GotoIf("SIP/test.3012-b7ebabe0", "1?s-LOCAL,1:s-NOTLOCAL,1") in new stack -- Goto (macro-dialdevice,s-LOCAL,1) -- Executing [s-LOCAL@macro-dialdevice:1] Dial("SIP/test.3012-b7ebabe0", "SIP/test.3011,20,r") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Audio is at 11.111.111.11 port 15800 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 22.222.222.22:42704: INVITE sip:test.3011@192.168.69.105:5062 SIP/2.0 Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a;rport Max-Forwards: 70 From: "Test" ;tag=as66fd48b9 To: Contact: Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.2-r212162 Date: Fri, 14 Aug 2009 16:10:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 837917842 837917842 IN IP4 11.111.111.11 s=Asterisk PBX SVN-branch-1.6.2-r212162 c=IN IP4 11.111.111.11 t=0 0 m=audio 15800 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called test.3011 pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> SIP/2.0 100 Trying To: From: "Test" ;tag=as66fd48b9 Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 INVITE Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a Server: Linksys/SPA962-5.2.8(SC) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> SIP/2.0 180 Ringing To: ;tag=6cf667bc579f1afci2 From: "Test" ;tag=as66fd48b9 Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 INVITE Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a Server: Linksys/SPA962-5.2.8(SC) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/test.3011-096698c0 is ringing -- Nobody picked up in 20000 ms Scheduling destruction of SIP dialog '549d17046255843d4c55126a2321007f@11.111.111.11' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 22.222.222.22:42704: CANCEL sip:test.3011@192.168.69.105:5062 SIP/2.0 Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a;rport Max-Forwards: 70 From: "Test" ;tag=as66fd48b9 To: Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 CANCEL User-Agent: Asterisk PBX SVN-branch-1.6.2-r212162 Content-Length: 0 --- Scheduling destruction of SIP dialog '549d17046255843d4c55126a2321007f@11.111.111.11' in 32000 ms (Method: INVITE) -- Executing [s-LOCAL@macro-dialdevice:2] MacroExit("SIP/test.3012-b7ebabe0", "") in new stack -- Executing [s@macro-stdexten:16] Set("SIP/test.3012-b7ebabe0", "box=3011") in new stack -- Executing [s@macro-stdexten:17] GotoIf("SIP/test.3012-b7ebabe0", "0?novm:hasvm") in new stack -- Goto (macro-stdexten,s,2000) -- Executing [s@macro-stdexten:2000] NoOp("SIP/test.3012-b7ebabe0", "Yeah Voicemail") in new stack -- Executing [s@macro-stdexten:2001] Goto("SIP/test.3012-b7ebabe0", "vmdial-NOANSWER,1") in new stack -- Goto (macro-stdexten,vmdial-NOANSWER,1) -- Executing [vmdial-NOANSWER@macro-stdexten:1] Goto("SIP/test.3012-b7ebabe0", "s,vmunavail") in new stack -- Goto (macro-stdexten,s,2002) -- Executing [s@macro-stdexten:2002] VoiceMail("SIP/test.3012-b7ebabe0", "3011@test,u") in new stack -- Playing '/var/spool/asterisk/voicemail/test/3011/greet.gsm' (language 'en') pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> SIP/2.0 487 Request Terminated To: ;tag=6cf667bc579f1afci2 From: "Test" ;tag=as66fd48b9 Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 INVITE Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a Server: Linksys/SPA962-5.2.8(SC) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 22.222.222.22:42704: ACK sip:test.3011@192.168.69.105:5062 SIP/2.0 Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a;rport Max-Forwards: 70 From: "Test" ;tag=as66fd48b9 To: ;tag=6cf667bc579f1afci2 Contact: Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.2-r212162 Content-Length: 0 pbx1*CLI> --- pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> SIP/2.0 200 OK To: ;tag=6cf667bc579f1afci2 From: "Test" ;tag=as66fd48b9 Call-ID: 549d17046255843d4c55126a2321007f@11.111.111.11 CSeq: 102 CANCEL Via: SIP/2.0/UDP 11.111.111.11:5060;branch=z9hG4bK6e73853a Server: Linksys/SPA962-5.2.8(SC) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '549d17046255843d4c55126a2321007f@11.111.111.11' Method: INVITE -- Playing 'vm-isunavail.ulaw' (language 'en') -- Playing 'vm-intro.ulaw' (language 'en') -- Playing 'beep.ulaw' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/test/3011/tmp/00C2lw format: wav49, 0x966aa50 -- x=1, open writing: /var/spool/asterisk/voicemail/test/3011/tmp/00C2lw format: gsm, 0x96756a0 -- x=2, open writing: /var/spool/asterisk/voicemail/test/3011/tmp/00C2lw format: wav, 0x9443ed8 -- User hung up == Parsing '/var/spool/asterisk/voicemail/test/3011/INBOX/msg0009.txt': == Found == Spawn extension (macro-stdexten, s, 2002) exited non-zero on 'SIP/test.3012-b7ebabe0' in macro 'stdexten' == Spawn extension (test#dial, 3011, 1) exited non-zero on 'SIP/test.3012-b7ebabe0' pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> REGISTER sip:pbx1.starnetworks.us SIP/2.0 Via: SIP/2.0/UDP 192.168.69.105:5062;branch=z9hG4bK-cb6c7a03 From: "3011" ;tag=68de2fe455f82885o2 To: "3011" Call-ID: fd163fdc-ab429627@192.168.69.105 CSeq: 57503 REGISTER Max-Forwards: 70 Authorization: Digest username="test.3011",realm="pbx1",nonce="661340fa",uri="sip:pbx1.starnetworks.us",algorithm=MD5,response="de19b62c5b4c1c32d9d93008546b17b1" Contact: "3011" ;expires=300 User-Agent: Linksys/SPA962-5.2.8(SC) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (13 headers 0 lines) --- <--- Transmitting (NAT) to 22.222.222.22:42704 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.69.105:5062;branch=z9hG4bK-cb6c7a03;received=22.222.222.22 From: "3011" ;tag=68de2fe455f82885o2 To: "3011" ;tag=as420b5372 Call-ID: fd163fdc-ab429627@192.168.69.105 CSeq: 57503 REGISTER Server: Asterisk PBX SVN-branch-1.6.2-r212162 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="pbx1", nonce="451ee4de" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'fd163fdc-ab429627@192.168.69.105' in 32000 ms (Method: REGISTER) pbx1*CLI> <--- SIP read from UDP:22.222.222.22:42704 ---> REGISTER sip:pbx1.starnetworks.us SIP/2.0 Via: SIP/2.0/UDP 192.168.69.105:5062;branch=z9hG4bK-25806cff From: "3011" ;tag=68de2fe455f82885o2 To: "3011" Call-ID: fd163fdc-ab429627@192.168.69.105 CSeq: 57504 REGISTER Max-Forwards: 70 Authorization: Digest username="test.3011",realm="pbx1",nonce="451ee4de",uri="sip:pbx1.starnetworks.us",algorithm=MD5,response="cd5372f1848523aa188fc9b4e6fafebc" Contact: "3011" ;expires=300 User-Agent: Linksys/SPA962-5.2.8(SC) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (13 headers 0 lines) --- Sending to 22.222.222.22 : 42704 (NAT) pbx1*CLI> <--- Transmitting (NAT) to 22.222.222.22:42704 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.69.105:5062;branch=z9hG4bK-25806cff;received=22.222.222.22 From: "3011" ;tag=68de2fe455f82885o2 To: "3011" ;tag=as420b5372 Call-ID: fd163fdc-ab429627@192.168.69.105 CSeq: 57504 REGISTER Server: Asterisk PBX SVN-branch-1.6.2-r212162 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 300 Contact: ;expires=300 Date: Fri, 14 Aug 2009 16:12:06 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'fd163fdc-ab429627@192.168.69.105' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'fd163fdc-ab429627@192.168.69.105' Method: REGISTER pbx1*CLI>