<-- 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=z9hG4bKc0a8006b0000001045c82f6300001e9200000a7f Content-Length: 0 Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107 CSeq: 856 OPTIONS From: ;tag=30468348116720 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=z9hG4bKc0a8006b0000001045c82f6300001e9200000a7f;received=24.7.133.237;rport=32767 From: ;tag=30468348116720 To: ;tag=as46d58896 Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107 CSeq: 856 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: OPTIONS sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c82f5100004e86000002e2 Content-Length: 0 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 206 OPTIONS From: ;tag=86210098413065 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=z9hG4bKc0a8006a0000001045c82f5100004e86000002e2;received=24.7.133.237;rport=63464 From: ;tag=86210098413065 To: ;tag=as2bbc15f7 Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106 CSeq: 206 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=z9hG4bK4c8c.5947a85.0 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024 Max-Forwards: 16 From: ;tag=a069d4400018ad2643bb06040e08b79f To: Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 3228263589-436015572-2804583647-138976334 h323-conf-id: 3228263589-436015572-2804583647-138976334 Content-Length: 357 Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 4114 1455 IN IP4 204.147.183.18 s=SIP Call i=oldc:c=IN IP4 66.114.240.12 t=0 0 m=audio 58924 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 - C07353DE-19FD11D4-A72D8CDF-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 204.147.183.18:58924 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: SIP/2.0 100 Trying Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024 From: ;tag=a069d4400018ad2643bb06040e08b79f To: Call-ID: C07353DE-19FD11D4-A72D8CDF-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-b640e6b0", "") 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=z9hG4bK4c8c.5947a85.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024 From: ;tag=a069d4400018ad2643bb06040e08b79f To: ;tag=as650b83d7 Call-ID: C07353DE-19FD11D4-A72D8CDF-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-b640e6b0", "1?from-trunk||1") in new stack -- Goto (from-trunk,s,1) -- Executing Set("SIP/08918917-b640e6b0", "FROM_DID=s") in new stack -- Executing Goto("SIP/08918917-b640e6b0", "ext-local|1234|1") in new stack -- Goto (ext-local,1234,1) -- Executing Macro("SIP/08918917-b640e6b0", "exten-vm|1234|1234") in new stack -- Executing Macro("SIP/08918917-b640e6b0", "user-callerid") in new stack -- Executing NoOp("SIP/08918917-b640e6b0", "user-callerid: 9166067933") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "0?report") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "0?start") in new stack -- Executing Set("SIP/08918917-b640e6b0", "REALCALLERIDNUM=9166067933") in new stack -- Executing NoOp("SIP/08918917-b640e6b0", "REALCALLERIDNUM is 9166067933") in new stack -- Executing Set("SIP/08918917-b640e6b0", "AMPUSER=") in new stack -- Executing Set("SIP/08918917-b640e6b0", "AMPUSERCIDNAME=") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "1?report") in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp("SIP/08918917-b640e6b0", "TTL: ARG1: 1234") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "0?continue") in new stack -- Executing Set("SIP/08918917-b640e6b0", "_TTL=64") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/08918917-b640e6b0", "Using CallerID "" <9166067933>") in new stack -- Executing Set("SIP/08918917-b640e6b0", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/08918917-b640e6b0", "VMBOX=1234") in new stack -- Executing Set("SIP/08918917-b640e6b0", "EXTTOCALL=1234") in new stack -- Executing Set("SIP/08918917-b640e6b0", "CFUEXT=") in new stack -- Executing Set("SIP/08918917-b640e6b0", "RT=55") in new stack -- Executing Macro("SIP/08918917-b640e6b0", "record-enable|1234|IN") in new stack -- Executing GotoIf("SIP/08918917-b640e6b0", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/08918917-b640e6b0", "recordingcheck|20070206-103316|1170775996.40") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070206-103316|1170775996.40: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/08918917-b640e6b0", "No recording needed") in new stack -- Executing Macro("SIP/08918917-b640e6b0", "dial|55|trw|1234") in new stack -- Executing DeadAGI("SIP/08918917-b640e6b0", "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-b640e6b0", "SIP/1234|55|trw") in new stack We're at 38.103.144.10 port 17142 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=z9hG4bK3dd64247;rport From: "9166067933" ;tag=as3255c8d8 To: Contact: Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 06 Feb 2007 15:33:16 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 17142 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=z9hG4bK4c8c.5947a85.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024 From: ;tag=a069d4400018ad2643bb06040e08b79f To: ;tag=as650b83d7 Call-ID: C07353DE-19FD11D4-A72D8CDF-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=z9hG4bK3dd64247 Content-Length: 0 Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10 CSeq: 102 INVITE From: "9166067933";tag=as3255c8d8 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=8621074843830 --- (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=z9hG4bK3dd64247 Content-Length: 0 Contact: Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10 CSeq: 102 INVITE From: "9166067933";tag=as3255c8d8 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=8621074843830 --- (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=z9hG4bK3dd64247 Content-Length: 219 Contact: Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10 Content-Type: application/sdp CSeq: 102 INVITE From: "9166067933";tag=as3255c8d8 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=8621074843830 v=0 o=- 3379736023 3379736023 IN IP4 24.7.133.237 s=SJphone c=IN IP4 24.7.133.237 t=0 0 a=direction:active m=audio 49182 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 24.7.133.237:49182 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=z9hG4bK64893096;rport From: "9166067933" ;tag=as3255c8d8 To: ;tag=8621074843830 Contact: Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1234-081e8110 answered SIP/08918917-b640e6b0 We're at 38.103.144.10 port 18144 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=z9hG4bK4c8c.5947a85.0;received=204.147.183.18 Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024 Record-Route: From: ;tag=a069d4400018ad2643bb06040e08b79f To: ;tag=as650b83d7 Call-ID: C07353DE-19FD11D4-A72D8CDF-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 18144 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=z9hG4bK2cc3e5643859418549c0530ea87ab971 Max-Forwards: 16 From: ;tag=a069d4400018ad2643bb06040e08b79f To: ;tag=as650b83d7 Call-ID: C07353DE-19FD11D4-A72D8CDF-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: REGISTER sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81 Content-Length: 0 Contact: Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 291 REGISTER From: ;tag=3046947271721 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 : 32767 (NAT) Transmitting (NAT) to 24.7.133.237:32767: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81;received=24.7.133.237;rport=32767 From: ;tag=3046947271721 To: Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 291 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --1*CLI> Transmitting (NAT) to 24.7.133.237:32767: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81;received=24.7.133.237;rport=32767 From: ;tag=3046947271721 To: ;tag=as7cd9134c Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 291 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a7e46be" Content-Length: 0 --- Scheduling destruction of call '854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107' in 15000 ms PP1*CLI> <-- SIP read from 24.7.133.237:32767: REGISTER sip:38.103.144.10:5060 SIP/2.0 Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84 Content-Length: 0 Contact: Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 292 REGISTER From: ;tag=30469481717155 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username="2134",realm="asterisk",nonce="7a7e46be",uri="sip:38.103.144.10:5060",response="038a338e25e22cc5d7fa58e95b4881d6",algorithm="MD5" --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 24.7.133.237 : 32767 (NAT) Transmitting (NAT) to 24.7.133.237:32767: SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84;received=24.7.133.237;rport=32767 From: ;tag=30469481717155 To: Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 292 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 ---*CLI> Transmitting (NAT) to 24.7.133.237:32767: SIP/2.0 200 OK Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84;received=24.7.133.237;rport=32767 From: ;tag=30469481717155 To: ;tag=as7cd9134c Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107 CSeq: 292 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Tue, 06 Feb 2007 15:33:21 GMT Content-Length: 0 --- Scheduling destruction of call '854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107' in 15000 ms