ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> [Mar 14 19:06:57] RTP-stats [Mar 14 19:06:57] * Our Receiver: [Mar 14 19:06:57] SSRC: 0 [Mar 14 19:06:57] Received packets: 0 [Mar 14 19:06:57] Lost packets: 0 [Mar 14 19:06:57] Jitter: 0.0000 [Mar 14 19:06:57] Transit: 0.0000 [Mar 14 19:06:57] RR-count: 0 [Mar 14 19:06:57] * Our Sender: [Mar 14 19:06:57] SSRC: 983644484 [Mar 14 19:06:57] Sent packets: 0 [Mar 14 19:06:57] Lost packets: 0 [Mar 14 19:06:57] Jitter: 0 [Mar 14 19:06:57] SR-count: 0 [Mar 14 19:06:57] RTT: 0.000000 [Mar 14 19:06:57] Audio is at 167.206.178.7 port 15960 [Mar 14 19:06:57] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:06:57] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:06:57] Reliably Transmitting (no NAT) to 69.16.233.35:5060: INVITE sip:15083282553@69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d43c6ce;rport From: "6025161069" ;tag=as32d99ae1 To: Contact: Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:06:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 27616 27616 IN IP4 167.206.178.7 s=session c=IN IP4 167.206.178.7 t=0 0 m=audio 15960 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 --- [Mar 14 19:06:57] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d43c6ce;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as32d99ae1 To: Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> [Mar 14 19:06:57] --- (10 headers 0 lines) --- [Mar 14 19:07:02] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d43c6ce;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as32d99ae1 To: ;tag=as1c3e1289 Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 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 16631 16631 IN IP4 69.16.233.35 s=session c=IN IP4 69.16.233.35 t=0 0 m=audio 15688 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Mar 14 19:07:02] --- (11 headers 10 lines) --- [Mar 14 19:07:02] Found RTP audio format 0 [Mar 14 19:07:02] Found RTP audio format 101 [Mar 14 19:07:02] Peer audio RTP is at port 69.16.233.35:15688 [Mar 14 19:07:02] Found description format PCMU for ID 0 [Mar 14 19:07:02] Found description format telephone-event for ID 101 [Mar 14 19:07:02] Got unsupported a:fmtp in SDP offer [Mar 14 19:07:02] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:07:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:07:02] Peer audio RTP is at port 69.16.233.35:15688 [Mar 14 19:07:05] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d43c6ce;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as32d99ae1 To: ;tag=as1c3e1289 Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 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 16631 16632 IN IP4 69.16.233.35 s=session c=IN IP4 69.16.233.35 t=0 0 m=audio 15688 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Mar 14 19:07:05] --- (11 headers 10 lines) --- [Mar 14 19:07:05] Found RTP audio format 0 [Mar 14 19:07:05] Found RTP audio format 101 [Mar 14 19:07:05] Peer audio RTP is at port 69.16.233.35:15688 [Mar 14 19:07:05] Found description format PCMU for ID 0 [Mar 14 19:07:05] Found description format telephone-event for ID 101 [Mar 14 19:07:05] Got unsupported a:fmtp in SDP offer [Mar 14 19:07:05] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:07:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:07:05] Peer audio RTP is at port 69.16.233.35:15688 [Mar 14 19:07:05] list_route: hop: [Mar 14 19:07:05] set_destination: Parsing for address/port to send to [Mar 14 19:07:05] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:07:05] Transmitting (no NAT) to 69.16.233.35:5060: ACK sip:15083282553@69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6bdc7f94;rport From: "6025161069" ;tag=as32d99ae1 To: ;tag=as1c3e1289 Contact: Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:07:05] > Channel SIP/voipjet-00708330 was answered. [Mar 14 19:07:05] -- Executing [s@liveonly:1] Answer("SIP/voipjet-00708330", "") in new stack [Mar 14 19:07:05] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-00708330", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:481 CALLERID: ") in new stack [Mar 14 19:07:05] -- Executing [s@liveonly:3] Wait("SIP/voipjet-00708330", "1") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:4] Set("SIP/voipjet-00708330", "soundfile=DEBUG-20070314190706-1173913617.47.wav") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-00708330", "DEBUG-20070314190706-1173913617.47.wav") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:6] Set("SIP/voipjet-00708330", "CDR(accountcode)=481") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:7] Set("SIP/voipjet-00708330", "CDR(userfield)= 5083282553") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-00708330", "machinedetect") in new stack [Mar 14 19:07:06] -- Executing [s@liveonly:9] AMD("SIP/voipjet-00708330", "") in new stack [Mar 14 19:07:06] -- AMD: SIP/voipjet-00708330 6025161069 (null) (Fmt: 4) [Mar 14 19:07:06] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 19:07:07] -- AMD: Word detected. iWordsCount:1 [Mar 14 19:07:08] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 19:07:09] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 19:07:09] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-00708330", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 19:07:09] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-00708330", "0?liveonly-amd|s|1") in new stack [Mar 14 19:07:09] -- Executing [s@liveonly:12] Goto("SIP/voipjet-00708330", "liveonly-live|s|1") in new stack [Mar 14 19:07:09] -- Goto (liveonly-live,s,1) [Mar 14 19:07:09] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-00708330", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 19:07:09] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 19:07:11] == CDR updated on SIP/voipjet-00708330 [Mar 14 19:07:11] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-00708330", "Press 1 transfer") in new stack [Mar 14 19:07:11] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-00708330", "liveonly-transfer|s|1") in new stack [Mar 14 19:07:11] -- Goto (liveonly-transfer,s,1) [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-00708330", "soundfile=20070314190711-1173913617.47.ulaw") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-00708330", "20070314190711-1173913617.47.ulaw|b") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CALLERID(num)=3024573281") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-00708330", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-00708330", "CDR(userfield5)=2007-03-14 19:07:11") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-00708330", "0?20") in new stack [Mar 14 19:07:11] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-00708330", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 19:07:11] == Begin MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:07:11] RTP-stats [Mar 14 19:07:11] * Our Receiver: [Mar 14 19:07:11] SSRC: 0 [Mar 14 19:07:11] Received packets: 0 [Mar 14 19:07:11] Lost packets: 0 [Mar 14 19:07:11] Jitter: 0.0000 [Mar 14 19:07:11] Transit: 0.0000 [Mar 14 19:07:11] RR-count: 0 [Mar 14 19:07:11] * Our Sender: [Mar 14 19:07:11] SSRC: 1288452152 [Mar 14 19:07:11] Sent packets: 0 [Mar 14 19:07:11] Lost packets: 0 [Mar 14 19:07:11] Jitter: 0 [Mar 14 19:07:11] SR-count: 0 [Mar 14 19:07:11] RTT: 0.000000 [Mar 14 19:07:11] Audio is at 167.206.178.7 port 18554 [Mar 14 19:07:11] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:07:11] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:07:11] Reliably Transmitting (no NAT) to 69.16.233.35:5060: INVITE sip:13024573281@69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK32d2f2b6;rport From: "3024573281" ;tag=as59cbf48f To: Contact: Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:07:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 27616 27616 IN IP4 167.206.178.7 s=session c=IN IP4 167.206.178.7 t=0 0 m=audio 18554 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 --- [Mar 14 19:07:11] -- Called 13024573281@voipjet [Mar 14 19:07:11] -- Started music on hold, class 'trans3', on SIP/voipjet-00708330 [Mar 14 19:07:12] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK32d2f2b6;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as59cbf48f To: Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> [Mar 14 19:07:12] --- (10 headers 0 lines) --- [Mar 14 19:07:14] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK32d2f2b6;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as59cbf48f To: ;tag=as2e3d5f77 Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 102 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 16631 16631 IN IP4 69.16.233.35 s=session c=IN IP4 69.16.233.35 t=0 0 m=audio 16662 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Mar 14 19:07:14] --- (11 headers 10 lines) --- [Mar 14 19:07:14] Found RTP audio format 0 [Mar 14 19:07:14] Found RTP audio format 101 [Mar 14 19:07:14] Peer audio RTP is at port 69.16.233.35:16662 [Mar 14 19:07:14] Found description format PCMU for ID 0 [Mar 14 19:07:14] Found description format telephone-event for ID 101 [Mar 14 19:07:14] Got unsupported a:fmtp in SDP offer [Mar 14 19:07:14] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:07:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:07:14] Peer audio RTP is at port 69.16.233.35:16662 [Mar 14 19:07:14] -- Call on SIP/voipjet-00726fa0 left from hold [Mar 14 19:07:14] -- Stopped music on hold on SIP/voipjet-00708330 [Mar 14 19:07:14] -- SIP/voipjet-00726fa0 is making progress passing it to SIP/voipjet-00708330 [Mar 14 19:07:17] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK32d2f2b6;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as59cbf48f To: ;tag=as2e3d5f77 Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 102 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 16631 16632 IN IP4 69.16.233.35 s=session c=IN IP4 69.16.233.35 t=0 0 m=audio 16662 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Mar 14 19:07:17] --- (11 headers 10 lines) --- [Mar 14 19:07:17] Found RTP audio format 0 [Mar 14 19:07:17] Found RTP audio format 101 [Mar 14 19:07:17] Peer audio RTP is at port 69.16.233.35:16662 [Mar 14 19:07:17] Found description format PCMU for ID 0 [Mar 14 19:07:17] Found description format telephone-event for ID 101 [Mar 14 19:07:17] Got unsupported a:fmtp in SDP offer [Mar 14 19:07:17] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:07:17] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:07:17] Peer audio RTP is at port 69.16.233.35:16662 [Mar 14 19:07:17] list_route: hop: [Mar 14 19:07:17] set_destination: Parsing for address/port to send to [Mar 14 19:07:17] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:07:17] Transmitting (no NAT) to 69.16.233.35:5060: ACK sip:13024573281@69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6759dfee;rport From: "3024573281" ;tag=as59cbf48f To: ;tag=as2e3d5f77 Contact: Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:07:17] -- Call on SIP/voipjet-00726fa0 left from hold [Mar 14 19:07:17] -- SIP/voipjet-00726fa0 answered SIP/voipjet-00708330 [Mar 14 19:07:32] Reliably Transmitting (no NAT) to 69.16.233.35:5060: OPTIONS sip:69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK3a0b6905;rport From: "asterisk" ;tag=as18f08ea6 To: Contact: Call-ID: 7f38b3162ae296810bd1a52c0378c19f@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:07:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:07:32] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK3a0b6905;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as18f08ea6 To: ;tag=as56e9db3a Call-ID: 7f38b3162ae296810bd1a52c0378c19f@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 <-------------> [Mar 14 19:07:32] --- (10 headers 0 lines) --- [Mar 14 19:07:32] Really destroying SIP dialog '7f38b3162ae296810bd1a52c0378c19f@167.206.178.7' Method: OPTIONS [Mar 14 19:07:33] Reliably Transmitting (no NAT) to 69.45.170.230:5060: OPTIONS sip:69.45.170.230 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2f48863a;rport From: "asterisk" ;tag=as6c263142 To: Contact: Call-ID: 4abbdaa35033858f227ce7e7052139db@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:07:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:07:33] <--- SIP read from 69.45.170.230:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2f48863a;rport From: "asterisk" ;tag=as6c263142 To: Call-ID: 4abbdaa35033858f227ce7e7052139db@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 19:07:33] --- (7 headers 0 lines) --- [Mar 14 19:07:33] Really destroying SIP dialog '4abbdaa35033858f227ce7e7052139db@167.206.178.7' Method: OPTIONS [Mar 14 19:07:33] Reliably Transmitting (no NAT) to 66.63.165.151:5060: OPTIONS sip:66.63.165.151 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2a8f420d;rport From: "asterisk" ;tag=as50542d21 To: Contact: Call-ID: 28c6af3f4889180168b258cb132a8bbe@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:07:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:07:33] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2a8f420d;rport=5060 From: "asterisk" ;tag=as50542d21 To: ;tag=0-tdb3221212128 Call-ID: 28c6af3f4889180168b258cb132a8bbe@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 19:07:33] --- (9 headers 0 lines) --- [Mar 14 19:07:33] Really destroying SIP dialog '28c6af3f4889180168b258cb132a8bbe@167.206.178.7' Method: OPTIONS [Mar 14 19:07:35] <--- SIP read from 69.16.233.35:5060 ---> BYE sip:6025161069@167.206.178.7 SIP/2.0 Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK4f375260;rport From: ;tag=as1c3e1289 To: "6025161069" ;tag=as32d99ae1 Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 19:07:35] --- (9 headers 0 lines) --- [Mar 14 19:07:35] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 19:07:35] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK4f375260;received=69.16.233.35;rport=5060 From: ;tag=as1c3e1289 To: "6025161069" ;tag=as32d99ae1 Call-ID: 278d9cc93914e9a0093d51034ab35cab@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 14 19:07:35] Scheduling destruction of SIP dialog '5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 19:07:35] set_destination: Parsing for address/port to send to [Mar 14 19:07:35] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:07:35] Reliably Transmitting (no NAT) to 69.16.233.35:5060: BYE sip:13024573281@69.16.233.35 SIP/2.0 Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK773919bd;rport From: "3024573281" ;tag=as59cbf48f To: ;tag=as2e3d5f77 Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:07:35] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-00708330' [Mar 14 19:07:35] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-00708330", "CDR(userfield)=press1transfer 20070314190711-1173913617.47.ulaw") in new stack [Mar 14 19:07:35] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-00708330", "CDR(userfield2)=5083282553") in new stack [Mar 14 19:07:35] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 19:07:35] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-00708330", "CDR(accountcode)=481") in new stack [Mar 14 19:07:35] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-00708330", "5083282553") in new stack [Mar 14 19:07:35] == End MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:07:35] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK773919bd;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as59cbf48f To: ;tag=as2e3d5f77 Call-ID: 5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing <-------------> [Mar 14 19:07:35] --- (11 headers 0 lines) --- [Mar 14 19:07:35] Really destroying SIP dialog '5e6c1ad64fed1ffe4701444270b0a8ef@167.206.178.7' Method: INVITE [Mar 14 19:07:35] RTP-stats [Mar 14 19:07:35] * Our Receiver: [Mar 14 19:07:35] SSRC: 1097947884 [Mar 14 19:07:35] Received packets: 1040 [Mar 14 19:07:35] Lost packets: 1 [Mar 14 19:07:35] Jitter: 0.0001 [Mar 14 19:07:35] Transit: -0.0001 [Mar 14 19:07:35] RR-count: 4 [Mar 14 19:07:35] * Our Sender: [Mar 14 19:07:35] SSRC: 1710775078 [Mar 14 19:07:35] Sent packets: 0 [Mar 14 19:07:35] Lost packets: 0 [Mar 14 19:07:35] Jitter: 0 [Mar 14 19:07:35] SR-count: 0 [Mar 14 19:07:35] RTT: 0.000000 [Mar 14 19:07:35] Really destroying SIP dialog '278d9cc93914e9a0093d51034ab35cab@167.206.178.7' Method: BYE [Mar 14 19:07:35] RTP-stats [Mar 14 19:07:35] * Our Receiver: [Mar 14 19:07:35] SSRC: 823113770 [Mar 14 19:07:35] Received packets: 1607 [Mar 14 19:07:35] Lost packets: 2 [Mar 14 19:07:35] Jitter: 0.0002 [Mar 14 19:07:35] Transit: 0.0002 [Mar 14 19:07:35] RR-count: 1 [Mar 14 19:07:35] * Our Sender: [Mar 14 19:07:35] SSRC: 1406973715 [Mar 14 19:07:35] Sent packets: 1131 [Mar 14 19:07:35] Lost packets: 0 [Mar 14 19:07:35] Jitter: 0 [Mar 14 19:07:35] SR-count: 5 [Mar 14 19:07:35] RTT: 0.000000