rendered paste bodyConnected to Asterisk 1.2.14 currently running on PP1 (pid = 4945)Verbosity was 4 and is now 11REGISTER 12 headers, 0 linesReliably Transmitting (no NAT) to 4.78.224.161:5060:REGISTER sip:4.78.224.161 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4fbc4e31;rportFrom: <sip:9048664201@4.78.224.161>;tag=as064a7989To: <sip:9048664201@4.78.224.161>Call-ID: 297d79511c30f6d93f261f453f60b098@127.0.0.1CSeq: 120 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:s@38.103.144.10>Event: registrationContent-Length: 0---PP1*CLI><-- SIP read from 4.78.224.161:5060:SIP/2.0 200 OKFrom: <sip:9048664201@4.78.224.161>;tag=as064a7989To: <sip:9048664201@4.78.224.161>;tag=192.168.67.2+1+0+cab13640Server: DC-SIP/2.0Organization:Content-Length: 0Call-ID: 297d79511c30f6d93f261f453f60b098@127.0.0.1CSeq: 120 REGISTERVia: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK4fbc4e31Contact: sip:s@4.78.224.161:5060;expires=3600--- (10 headers 0 lines) ---Scheduling destruction of call '297d79511c30f6d93f261f453f60b098@127.0.0.1' in 32000 msPP1*CLI><-- SIP read from 24.7.133.237:63464:OPTIONS sip:38.103.144.10:5060 SIP/2.0Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c831fa00006ec200000345Content-Length: 0Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106CSeq: 239 OPTIONSFrom: <sip:1234@38.103.144.10:5060>;tag=8627823123909Max-Forwards: 70To: <sip:38.103.144.10:5060>--- (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 OKVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c831fa00006ec200000345;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=8627823123909To: <sip:38.103.144.10:5060>;tag=as4f37cc89Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106CSeq: 239 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:38.103.144.10>Accept: application/sdpContent-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/2.0Record-Route: <sip:204.147.183.18;ftag=007351d78d9deb01b174768bf7b51af9;lr=on>Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024Max-Forwards: 16From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9To: <sip:08918917@sip-ny1.stanaphone.com>Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 INVITEContact: Anonymous <sip:204.147.183.55:1024>Expires: 300User-Agent: Sippycisco-GUID: 1442768829-436146644-2817035487-138976334h323-conf-id: 1442768829-436146644-2817035487-138976334Content-Length: 357Content-Type: application/sdpv=0o=CiscoSystemsSIP-GW-UserAgent 6234 6714 IN IP4 204.147.183.18s=SIP Calli=oldc:c=IN IP4 66.114.240.12t=0 0m=audio 59040 RTP/AVP 0 18 3 101c=IN IP4 204.147.183.18a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=yesa=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=direction:passivea=nortpproxy:yes--- (16 headers 15 lines) ---Using INVITE request as basis request - 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12Sending to 204.147.183.18 : 5060 (non-NAT)Found peer 'stanap'Found RTP audio format 0Found RTP audio format 18Found RTP audio format 3Found RTP audio format 101Peer audio RTP is at port 204.147.183.18:59040Found description format PCMUFound description format G729Found description format GSMFound description format telephone-eventCapabilities: 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: <sip:204.147.183.18;ftag=007351d78d9deb01b174768bf7b51af9;lr=on>Transmitting (no NAT) to 204.147.183.18:5060:IP/2.0 100 TryingVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9To: <sip:08918917@sip-ny1.stanaphone.com>Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0--- -- Executing Ringing("SIP/08918917-b6416ea8", "") in new stackTransmitting (no NAT) to 204.147.183.18:5060:SIP/2.0 180 RingingVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9o: <sip:08918917@sip-ny1.stanaphone.com>;tag=as68612d90Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>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 stackWe're at 38.103.144.10 port 10368Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDP13 headers, 11 linesReliably Transmitting (NAT) to 24.7.133.237:63464:INVITE sip:1234@24.7.133.237:63464 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK00b51cd5;rportFrom: "9166067933" <sip:9166067933@38.103.144.10>;tag=as1583e348To: <sip:1234@24.7.133.237:63464>Contact: <sip:9166067933@38.103.144.10>Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Tue, 06 Feb 2007 15:44:36 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 240v=0o=root 4945 4945 IN IP4 38.103.144.10s=sessionc=IN IP4 38.103.144.10t=0 0m=audio 10368 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ---- -- Called 1234Transmitting (no NAT) to 204.147.183.18:5060:SIP/2.0 180 RingingVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as68612d90Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Length: 0---PP1*CLI><-- SIP read from 24.7.133.237:63464:SIP/2.0 100 TryingVia: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5Content-Length: 0Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 102 INVITEFrom: "9166067933"<sip:9166067933@38.103.144.10>;tag=as1583e348Server: SJphone/1.60.289a (SJ Labs)To: "unknown"<sip:1234@24.7.133.237:63464>;tag=86278792132066--- (8 headers 0 lines) ---PP1*CLI><-- SIP read from 24.7.133.237:63464:SIP/2.0 180 RingingVia: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5Content-Length: 0Contact: <sip:1234@24.7.133.237:63464>Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 102 INVITEFrom: "9166067933"<sip:9166067933@38.103.144.10>;tag=as1583e348Server: SJphone/1.60.289a (SJ Labs)To: "unknown"<sip:1234@24.7.133.237:63464>;tag=86278792132066--- (9 headers 0 lines) --- -- SIP/1234-081e8110 is ringingPP1*CLI><-- SIP read from 24.7.133.237:63464:SIP/2.0 200 OKVia: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK00b51cd5Content-Length: 219Contact: <sip:1234@24.7.133.237:63464>Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10Content-Type: application/sdpCSeq: 102 INVITEFrom: "9166067933"<sip:9166067933@38.103.144.10>;tag=as1583e348Server: SJphone/1.60.289a (SJ Labs)To: "unknown"<sip:1234@24.7.133.237:63464>;tag=86278792132066v=0o=- 3379736704 3379736704 IN IP4 24.7.133.237s=SJphonec=IN IP4 24.7.133.237t=0 0a=direction:activem=audio 49184 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,16--- (10 headers 10 lines) ---Found RTP audio format 0Found RTP audio format 101Peer audio RTP is at port 24.7.133.237:49184Found description format PCMUFound description format telephone-eventCapabilities: 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: <sip:1234@24.7.133.237:63464>set_destination: Parsing <sip:1234@24.7.133.237:63464> for address/port to send toset_destination: set destination to 24.7.133.237, port 63464Transmitting (NAT) to 24.7.133.237:63464:ACK sip:1234@24.7.133.237:63464 SIP/2.0Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5a0463db;rportFrom: "9166067933" <sip:9166067933@38.103.144.10>;tag=as1583e348To: <sip:1234@24.7.133.237:63464>;tag=86278792132066Contact: <sip:9166067933@38.103.144.10>Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0--- -- SIP/1234-081e8110 answered SIP/08918917-b6416ea8We're at 38.103.144.10 port 10216Adding codec 0x4 (ulaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 204.147.183.18:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bKcd3a.c66526d7.0;received=204.147.183.18Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bKc9e6fdb9a4e0262773d691d4fce52766;rport=1024Record-Route: <sip:204.147.183.18;ftag=007351d78d9deb01b174768bf7b51af9;lr=on>From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as68612d90Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:s@38.103.144.10>Content-Type: application/sdpContent-Length: 216v=0o=root 4945 4945 IN IP4 38.103.144.10s=sessionc=IN IP4 38.103.144.10t=0 0m=audio 10216 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ----PP1*CLI><-- SIP read from 204.147.183.18:5060:ACK sip:s@204.147.183.18:5060 SIP/2.0Via: SIP/2.0/UDP 204.147.183.18;branch=0Via: SIP/2.0/UDP 204.147.183.55:1024;rport=1024;branch=z9hG4bKf66a952e7c98415c9870c053de556dbcMax-Forwards: 16From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=007351d78d9deb01b174768bf7b51af9To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as68612d90Call-ID: 5603C995-19FF11D4-A7EB8CDF-8489C4E@66.114.240.12CSeq: 200 ACKExpires: 300User-Agent: Sippy--- (10 headers 0 lines) ---PP1*CLI><-- SIP read from 24.7.133.237:32767:OPTIONS sip:38.103.144.10:5060 SIP/2.0Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c8321500000eca00000ae5Content-Length: 0Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107CSeq: 889 OPTIONSFrom: <sip:2134@38.103.144.10:5060>;tag=30537299323798Max-Forwards: 70To: <sip:38.103.144.10:5060>--- (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 OKVia: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c8321500000eca00000ae5;received=24.7.133.237;rport=32767From: <sip:2134@38.103.144.10:5060>;tag=30537299323798To: <sip:38.103.144.10:5060>;tag=as51f7e1aaCall-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107CSeq: 889 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:38.103.144.10>Accept: application/sdpContent-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/2.0Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a000000ca45c8320c0000119300000349Content-Length: 0Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 1 BYEFrom: "unknown"<sip:1234@24.7.133.237:63464>;tag=86278792132066Max-Forwards: 70To: <sip:9166067933@38.103.144.10>;tag=as1583e348User-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 OKVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a000000ca45c8320c0000119300000349;received=24.7.133.237;rport=63464From: "unknown"<sip:1234@24.7.133.237:63464>;tag=86278792132066To: <sip:9166067933@38.103.144.10>;tag=as1583e348Call-ID: 0339c5f8676fb7055e57c4273688ebb2@38.103.144.10CSeq: 1 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:9166067933@38.103.144.10>ontent-Length: 0X-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 stackPP1*CLI><-- SIP read from 24.7.133.237:63464:REGISTER sip:38.103.144.10:5060 SIP/2.0Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034bContent-Length: 0Contact: <sip:1234@24.7.133.237:63464>Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 87 REGISTERFrom: <sip:1234@38.103.144.10:5060>;tag=86280092132514Max-Forwards: 70To: <sip:1234@38.103.144.10:5060>User-Agent: SJphone/1.60.289a (SJ Labs)--- (10 headers 0 lines) ---Using latest REGISTER request as basis requestSending to 24.7.133.237 : 63464 (NAT)Transmitting (NAT) to 24.7.133.237:63464:SIP/2.0 100 TryingVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034b;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=86280092132514To: <sip:1234@38.103.144.10:5060>Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 87 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:1234@38.103.144.10>Content-Length: 0---Transmitting (NAT) to 24.7.133.237:63464:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d00005ec30000034b;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=86280092132514To: <sip:1234@38.103.144.10:5060>;tag=as301564f4Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 87 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYWWW-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 msDestroying call '0339c5f8676fb7055e57c4273688ebb2@38.103.144.10'PP1*CLI><-- SIP read from 24.7.133.237:63464:REGISTER sip:38.103.144.10:5060 SIP/2.0Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034eContent-Length: 0Contact: <sip:1234@24.7.133.237:63464>Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 88 REGISTERFrom: <sip:1234@38.103.144.10:5060>;tag=8628010314545Max-Forwards: 70To: <sip:1234@38.103.144.10:5060>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 requestSending to 24.7.133.237 : 63464 (NAT)Transmitting (NAT) to 24.7.133.237:63464:SIP/2.0 100 TryingVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034e;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=8628010314545To: <sip:1234@38.103.144.10:5060>Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 88 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:1234@38.103.144.10>Content-Length: 0---Transmitting (NAT) to 24.7.133.237:63464:SIP/2.0 200 OKVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320d000062b60000034e;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=8628010314545To: <sip:1234@38.103.144.10:5060>;tag=as301564f4Call-ID: 5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106CSeq: 88 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYExpires: 120Contact: <sip:1234@24.7.133.237:63464>;expires=120Date: Tue, 06 Feb 2007 15:44:49 GMTContent-Length: 0---Scheduling destruction of call '5CC735D1-E8DF-48FD-A96F-3129F14CB04B@192.168.0.106' in 15000 msPP1*CLI><-- SIP read from 24.7.133.237:63464:OPTIONS sip:38.103.144.10:5060 SIP/2.0Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c8320e00004df300000351Content-Length: 0Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106CSeq: 240 OPTIONSFrom: <sip:1234@38.103.144.10:5060>;tag=86280231220474Max-Forwards: 70To: <sip:38.103.144.10:5060>--- (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 OKVia: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c8320e00004df300000351;received=24.7.133.237;rport=63464From: <sip:1234@38.103.144.10:5060>;tag=86280231220474To: <sip:38.103.144.10:5060>;tag=as4008a098Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106CSeq: 240 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:38.103.144.10>Accept: application/sdpContent-Length: 0---Destroying call 'C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106'PP1*CLI>