Connected to Asterisk 1.2.14 currently running on PP1 (pid = 4945) Verbosity was 4 and is now 11 REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4fbc4e31;rport From: ;tag=as064a7989 To: Call-ID: 297d79511c30f6d93f261f453f60b098@127.0.0.1 CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 200 OK From: ;tag=as064a7989 To: ;tag=192.168.67.2+1+0+cab13640 Server: DC-SIP/2.0 Organization: Content-Length: 0 Call-ID: 297d79511c30f6d93f261f453f60b098@127.0.0.1 CSeq: 120 REGISTER Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK4fbc4e31 Contact: sip:s@4.78.224.161:5060;expires=3600 --- (10 headers 0 lines) --- Scheduling destruction of call '297d79511c30f6d93f261f453f60b098@127.0.0.1' in 32000 ms PP1*CLI> <-- SIP read from 24.7.133.237:63464: OPTIONS sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c831fa00006ec200000345 Content-Length: 0 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 239 OPTIONS From: ;tag=8627823123909 Max-Forwards: 70 To: --- (8 headers 0 lines) --- Looking for s in from-sip-external (domain 38.103.144.10) Transmitting (no NAT) to 24.7.133.237:63464: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c831fa00006ec200000345;received=24.7.133.237;rport=63464 From: ;tag=8627823123909 To: ;tag=as4f37cc89 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 239 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106' PP1*CLI> <-- SIP read from 204.147.183.18:5060: INVITE sip:s@204.147.183.18:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024 Max-Forwards: 16 From: ;tag=007351d78d9deb01b174768bf7b51af9 To: Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 1442768829-436146644-2817035487-138976334 h323-conf-id: 1442768829-436146644-2817035487-138976334 Content-Length: 357 Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 6234 6714 IN IP4 204.147.183.18 s=SIP Call i=oldc:c=IN IP4 66.114.240.12 t=0 0 m=audio 59040 RTP/AVP 0 18 3 101 c=IN IP4 204.147.183.18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes --- (16 headers 15 lines) --- Using INVITE request as basis request - 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 Sending to 204.147.183.18 : 5060 (non-NAT) Found peer 'stanap' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 204.147.183.18:59040 Found description format PCMU Found description format G729 Found description format GSM Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in from-sip-external (domain 204.147.183.18) list_route: hop: Transmitting (no NAT) to 204.147.183.18:5060: IP/2.0 100 Trying Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024 From: ;tag=007351d78d9deb01b174768bf7b51af9 To: Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Ringing("SIP/08918917-b6416ea8", "") in new stack Transmitting (no NAT) to 204.147.183.18:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024 From: ;tag=007351d78d9deb01b174768bf7b51af9 o: ;tag=as68612d90 Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing GotoIf("SIP/08918917-b6416ea8", "1?from-trunk||1") in new stack -- Goto (from-trunk,s,1) -- Executing Set("SIP/08918917-b6416ea8", "FROM_DID=s") in new stack -- Executing Goto("SIP/08918917-b6416ea8", "ext-local|1234|1") in new stack -- Goto (ext-local,1234,1) -- Executing Macro("SIP/08918917-b6416ea8", "exten-vm|1234|1234") in new stack -- Executing Macro("SIP/08918917-b6416ea8", "user-callerid") in new stack -- Executing NoOp("SIP/08918917-b6416ea8", "user-callerid: 9166067933") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "0?report") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "0?start") in new stack -- Executing Set("SIP/08918917-b6416ea8", "REALCALLERIDNUM=9166067933") in new stack -- Executing NoOp("SIP/08918917-b6416ea8", "REALCALLERIDNUM is 9166067933") in new stack -- Executing Set("SIP/08918917-b6416ea8", "AMPUSER=") in new stack -- Executing Set("SIP/08918917-b6416ea8", "AMPUSERCIDNAME=") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "1?report") in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp("SIP/08918917-b6416ea8", "TTL: ARG1: 1234") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "0?continue") in new stack -- Executing Set("SIP/08918917-b6416ea8", "_TTL=64") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/08918917-b6416ea8", "Using CallerID "" <9166067933>") in new stack -- Executing Set("SIP/08918917-b6416ea8", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/08918917-b6416ea8", "VMBOX=1234") in new stack -- Executing Set("SIP/08918917-b6416ea8", "EXTTOCALL=1234") in new stack -- Executing Set("SIP/08918917-b6416ea8", "CFUEXT=") in new stack -- Executing Set("SIP/08918917-b6416ea8", "RT=55") in new stack -- Executing Macro("SIP/08918917-b6416ea8", "record-enable|1234|IN") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/08918917-b6416ea8", "recordingcheck|20070206-104436|1170776676.42") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070206-104436|1170776676.42: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/08918917-b6416ea8", "No recording needed") in new stack -- Executing Macro("SIP/08918917-b6416ea8", "dial|55|trw|1234") in new stack -- Executing DeadAGI("SIP/08918917-b6416ea8", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '9166067933' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 1234 to extension map -- dialparties.agi: Extension 1234 cf is disabled -- dialparties.agi: Extension 1234 do not disturb is disabled > dialparties.agi: extnum: 1234 > dialparties.agi: exthascw: 1 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: -- dialparties.agi: dbset CALLTRACE/1234 to 9166067933 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/08918917-b6416ea8", "SIP/1234|55|trw") in new stack We're at 38.103.144.10 port 10368 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (NAT) to 24.7.133.237:63464: INVITE sip:1234@24.7.133.237:63464 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK00b51cd5;rport From: "9166067933" ;tag=as1583e348 To: Contact: Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 06 Feb 2007 15:44:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 4945 4945 IN IP4 38.103.144.10 s=session c=IN IP4 38.103.144.10 t=0 0 m=audio 10368 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 1234 Transmitting (no NAT) to 204.147.183.18:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024 From: ;tag=007351d78d9deb01b174768bf7b51af9 To: ;tag=as68612d90 Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- PP1*CLI> <-- SIP read from 24.7.133.237:63464: SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5 Content-Length: 0 Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 102 INVITE From: "9166067933";tag=as1583e348 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=86278792132066 --- (8 headers 0 lines) --- PP1*CLI> <-- SIP read from 24.7.133.237:63464: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5 Content-Length: 0 Contact: Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 102 INVITE From: "9166067933";tag=as1583e348 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=86278792132066 --- (9 headers 0 lines) --- -- SIP/1234-081e8110 is ringing PP1*CLI> <-- SIP read from 24.7.133.237:63464: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5 Content-Length: 219 Contact: Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 Content-Type: application/sdp CSeq: 102 INVITE From: "9166067933";tag=as1583e348 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=86278792132066 v=0 o=- 3379736704 3379736704 IN IP4 24.7.133.237 s=SJphone c=IN IP4 24.7.133.237 t=0 0 a=direction:active m=audio 49184 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (10 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 24.7.133.237:49184 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 24.7.133.237, port 63464 Transmitting (NAT) to 24.7.133.237:63464: ACK sip:1234@24.7.133.237:63464 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5a0463db;rport From: "9166067933" ;tag=as1583e348 To: ;tag=86278792132066 Contact: Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1234-081e8110 answered SIP/08918917-b6416ea8 We're at 38.103.144.10 port 10216 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 204.147.183.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024 Record-Route: From: ;tag=007351d78d9deb01b174768bf7b51af9 To: ;tag=as68612d90 Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 4945 4945 IN IP4 38.103.144.10 s=session c=IN IP4 38.103.144.10 t=0 0 m=audio 10216 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- PP1*CLI> <-- SIP read from 204.147.183.18:5060: ACK sip:s@204.147.183.18:5060 SIP/2.0 Via: SIP/2.0/UDP 204.147.183.18;branch=0 Via: SIP/2.0/UDP 204.147.183.55:1024;rport=1024;branch=z9hG4bKf66a952e7c98415c9870c053de556dbc Max-Forwards: 16 From: ;tag=007351d78d9deb01b174768bf7b51af9 To: ;tag=as68612d90 Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12 CSeq: 200 ACK Expires: 300 User-Agent: Sippy --- (10 headers 0 lines) --- PP1*CLI> <-- SIP read from 24.7.133.237:32767: OPTIONS sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c8321500000eca00000ae5 Content-Length: 0 Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107 CSeq: 889 OPTIONS From: ;tag=30537299323798 Max-Forwards: 70 To: --- (8 headers 0 lines) --- Looking for s in from-sip-external (domain 38.103.144.10) Transmitting (no NAT) to 24.7.133.237:32767: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c8321500000eca00000ae5;received=24.7.133.237;rport=32767 From: ;tag=30537299323798 To: ;tag=as51f7e1aa Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107 CSeq: 889 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107' PP1*CLI> <-- SIP read from 24.7.133.237:63464: BYE sip:9166067933@38.103.144.10 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a000000ca45c8320c0000119300000349 Content-Length: 0 Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 1 BYE From: "unknown";tag=86278792132066 Max-Forwards: 70 To: ;tag=as1583e348 User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines) --- Sending to 24.7.133.237 : 63464 (NAT) Transmitting (NAT) to 24.7.133.237:63464: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a000000ca45c8320c0000119300000349;received=24.7.133.237;rport=63464 From: "unknown";tag=86278792132066 To: ;tag=as1583e348 Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: ontent-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/08918917-b6416ea8' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/08918917-b6416ea8' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/08918917-b6416ea8' -- Executing Macro("SIP/08918917-b6416ea8", "hangupcall") in new stack -- Executing ResetCDR("SIP/08918917-b6416ea8", "w") in new stack -- Executing NoCDR("SIP/08918917-b6416ea8", "") in new stack -- Executing GotoIf("SIP/08918917-b6416ea8", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/08918917-b6416ea8", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/08918917-b6416ea8", "5") in new stack PP1*CLI> <-- SIP read from 24.7.133.237:63464: REGISTER sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034b Content-Length: 0 Contact: Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 87 REGISTER From: ;tag=86280092132514 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 24.7.133.237 : 63464 (NAT) Transmitting (NAT) to 24.7.133.237:63464: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034b;received=24.7.133.237;rport=63464 From: ;tag=86280092132514 To: Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 24.7.133.237:63464: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034b;received=24.7.133.237;rport=63464 From: ;tag=86280092132514 To: ;tag=as301564f4 Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6497c2e9" Content-Length: 0 --- Scheduling destruction of call '5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106' in 15000 ms Destroying call '0339c5f8676fb7055e57c4273688ebb2@38.103.144.10' PP1*CLI> <-- SIP read from 24.7.133.237:63464: REGISTER sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034e Content-Length: 0 Contact: Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 88 REGISTER From: ;tag=8628010314545 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username="1234",realm="asterisk",nonce="6497c2e9",uri="sip:38.103.144.10:5060",response="faeb94a758a1b31ee206937a45c1d3e6",algorithm="MD5" --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 24.7.133.237 : 63464 (NAT) Transmitting (NAT) to 24.7.133.237:63464: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034e;received=24.7.133.237;rport=63464 From: ;tag=8628010314545 To: Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 24.7.133.237:63464: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034e;received=24.7.133.237;rport=63464 From: ;tag=8628010314545 To: ;tag=as301564f4 Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106 CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Tue, 06 Feb 2007 15:44:49 GMT Content-Length: 0 --- Scheduling destruction of call '5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106' in 15000 ms PP1*CLI> <-- SIP read from 24.7.133.237:63464: OPTIONS sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320e00004df300000351 Content-Length: 0 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 240 OPTIONS From: ;tag=86280231220474 Max-Forwards: 70 To: --- (8 headers 0 lines) --- Looking for s in from-sip-external (domain 38.103.144.10) Transmitting (no NAT) to 24.7.133.237:63464: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320e00004df300000351;received=24.7.133.237;rport=63464 From: ;tag=86280231220474 To: ;tag=as4008a098 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 240 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106' PP1*CLI>