[Mar 14 19:03:07] == Parsing '/etc/asterisk/manager.conf': [Mar 14 19:03:07] Found 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 19:03:26] RTP-stats [Mar 14 19:03:26] * Our Receiver: [Mar 14 19:03:26] SSRC: 0 [Mar 14 19:03:26] Received packets: 0 [Mar 14 19:03:26] Lost packets: 0 [Mar 14 19:03:26] Jitter: 0.0000 [Mar 14 19:03:26] Transit: 0.0000 [Mar 14 19:03:26] RR-count: 0 [Mar 14 19:03:26] * Our Sender: [Mar 14 19:03:26] SSRC: 1544545349 [Mar 14 19:03:26] Sent packets: 0 [Mar 14 19:03:26] Lost packets: 0 [Mar 14 19:03:26] Jitter: 0 [Mar 14 19:03:26] SR-count: 0 [Mar 14 19:03:26] RTT: 0.000000 [Mar 14 19:03:26] Audio is at 167.206.178.7 port 18746 [Mar 14 19:03:26] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:03:26] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:03:26] 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=z9hG4bK2f8234ba;rport From: "6025161069" ;tag=as70c79f3a To: Contact: Call-ID: 26e9cc1668f3e45520324f28541304b2@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:03:26 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 18746 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 19:03:26] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2f8234ba;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as70c79f3a To: Call-ID: 26e9cc1668f3e45520324f28541304b2@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 19:03:26] --- (10 headers 0 lines) --- [Mar 14 19:03:32] 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=z9hG4bK3caf8632;rport From: "asterisk" ;tag=as5f136d4d To: Contact: Call-ID: 4c5b396b7ec9c14d5fe6f3bd0f22dd48@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:03:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:03:32] <--- 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=z9hG4bK3caf8632;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as5f136d4d To: ;tag=as7f655734 Call-ID: 4c5b396b7ec9c14d5fe6f3bd0f22dd48@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 19:03:32] --- (10 headers 0 lines) --- [Mar 14 19:03:32] Really destroying SIP dialog '4c5b396b7ec9c14d5fe6f3bd0f22dd48@167.206.178.7' Method: OPTIONS [Mar 14 19:03:32] <--- 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=z9hG4bK2f8234ba;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as70c79f3a To: ;tag=as619bae0b Call-ID: 26e9cc1668f3e45520324f28541304b2@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 13154 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 19:03:32] --- (11 headers 10 lines) --- [Mar 14 19:03:32] Found RTP audio format 0 [Mar 14 19:03:32] Found RTP audio format 101 [Mar 14 19:03:32] Peer audio RTP is at port 69.16.233.35:13154 [Mar 14 19:03:32] Found description format PCMU for ID 0 [Mar 14 19:03:32] Found description format telephone-event for ID 101 [Mar 14 19:03:32] Got unsupported a:fmtp in SDP offer [Mar 14 19:03:32] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:03:32] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:03:32] Peer audio RTP is at port 69.16.233.35:13154 [Mar 14 19:03:32] 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=z9hG4bK635192c1;rport From: "asterisk" ;tag=as78008fe1 To: Contact: Call-ID: 63839dba0d977add07776d8951b7ad76@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:03:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:03:32] <--- 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=z9hG4bK635192c1;rport From: "asterisk" ;tag=as78008fe1 To: Call-ID: 63839dba0d977add07776d8951b7ad76@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 19:03:32] --- (7 headers 0 lines) --- [Mar 14 19:03:32] Really destroying SIP dialog '63839dba0d977add07776d8951b7ad76@167.206.178.7' Method: OPTIONS [Mar 14 19:03:33] 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=z9hG4bK2aa6c8bf;rport From: "asterisk" ;tag=as50e5963b To: Contact: Call-ID: 32e73d0927234dc353d8864e56650c0b@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:03:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:03:33] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2aa6c8bf;rport=5060 From: "asterisk" ;tag=as50e5963b To: ;tag=0-tdb3221212128 Call-ID: 32e73d0927234dc353d8864e56650c0b@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 19:03:33] --- (9 headers 0 lines) --- [Mar 14 19:03:33] Really destroying SIP dialog '32e73d0927234dc353d8864e56650c0b@167.206.178.7' Method: OPTIONS [Mar 14 19:03:38] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK2f8234ba;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as70c79f3a To: ;tag=as619bae0b Call-ID: 26e9cc1668f3e45520324f28541304b2@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 13154 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 19:03:38] --- (11 headers 10 lines) --- [Mar 14 19:03:38] Found RTP audio format 0 [Mar 14 19:03:38] Found RTP audio format 101 [Mar 14 19:03:38] Peer audio RTP is at port 69.16.233.35:13154 [Mar 14 19:03:38] Found description format PCMU for ID 0 [Mar 14 19:03:38] Found description format telephone-event for ID 101 [Mar 14 19:03:38] Got unsupported a:fmtp in SDP offer [Mar 14 19:03:38] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:03:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:03:38] Peer audio RTP is at port 69.16.233.35:13154 [Mar 14 19:03:38] list_route: hop: [Mar 14 19:03:38] set_destination: Parsing for address/port to send to [Mar 14 19:03:38] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:03:38] 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=z9hG4bK6a55c9b1;rport From: "6025161069" ;tag=as70c79f3a To: ;tag=as619bae0b Contact: Call-ID: 26e9cc1668f3e45520324f28541304b2@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:03:38] > Channel SIP/voipjet-00708330 was answered. [Mar 14 19:03:38] -- Executing [s@liveonly:1] Answer("SIP/voipjet-00708330", "") in new stack [Mar 14 19:03:38] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-00708330", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:479 CALLERID: ") in new stack [Mar 14 19:03:38] -- Executing [s@liveonly:3] Wait("SIP/voipjet-00708330", "1") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:4] Set("SIP/voipjet-00708330", "soundfile=DEBUG-20070314190340-1173913406.43.wav") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-00708330", "DEBUG-20070314190340-1173913406.43.wav") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:6] Set("SIP/voipjet-00708330", "CDR(accountcode)=479") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:7] Set("SIP/voipjet-00708330", "CDR(userfield)= 5083282553") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-00708330", "machinedetect") in new stack [Mar 14 19:03:40] -- Executing [s@liveonly:9] AMD("SIP/voipjet-00708330", "") in new stack [Mar 14 19:03:40] -- AMD: SIP/voipjet-00708330 6025161069 (null) (Fmt: 4) [Mar 14 19:03:40] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 19:03:40] -- AMD: Word detected. iWordsCount:1 [Mar 14 19:03:41] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 19:03:42] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 19:03:42] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-00708330", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 19:03:42] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-00708330", "0?liveonly-amd|s|1") in new stack [Mar 14 19:03:42] -- Executing [s@liveonly:12] Goto("SIP/voipjet-00708330", "liveonly-live|s|1") in new stack [Mar 14 19:03:42] -- Goto (liveonly-live,s,1) [Mar 14 19:03:42] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-00708330", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 19:03:42] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 19:03:49] == CDR updated on SIP/voipjet-00708330 [Mar 14 19:03:49] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-00708330", "Press 1 transfer") in new stack [Mar 14 19:03:49] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-00708330", "liveonly-transfer|s|1") in new stack [Mar 14 19:03:49] -- Goto (liveonly-transfer,s,1) [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-00708330", "soundfile=20070314190349-1173913406.43.ulaw") in new stack [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-00708330", "20070314190349-1173913406.43.ulaw|b") in new stack [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CALLERID(num)=3024573281") in new stack [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-00708330", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-00708330", "CDR(userfield5)=2007-03-14 19:03:49") in new stack [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-00708330", "0?20") in new stack [Mar 14 19:03:49] == Begin MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:03:49] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-00708330", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 19:03:49] RTP-stats [Mar 14 19:03:49] * Our Receiver: [Mar 14 19:03:49] SSRC: 0 [Mar 14 19:03:49] Received packets: 0 [Mar 14 19:03:49] Lost packets: 0 [Mar 14 19:03:49] Jitter: 0.0000 [Mar 14 19:03:49] Transit: 0.0000 [Mar 14 19:03:49] RR-count: 0 [Mar 14 19:03:49] * Our Sender: [Mar 14 19:03:49] SSRC: 850168249 [Mar 14 19:03:49] Sent packets: 0 [Mar 14 19:03:49] Lost packets: 0 [Mar 14 19:03:49] Jitter: 0 [Mar 14 19:03:49] SR-count: 0 [Mar 14 19:03:49] RTT: 0.000000 [Mar 14 19:03:49] Audio is at 167.206.178.7 port 16278 [Mar 14 19:03:49] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:03:49] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:03:49] 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=z9hG4bK4d6faf5e;rport From: "3024573281" ;tag=as35bd9002 To: Contact: Call-ID: 1ff878460fa6757e752e68a0704c27c9@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:03:49 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 16278 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 19:03:49] -- Called 13024573281@voipjet [Mar 14 19:03:49] -- Started music on hold, class 'trans3', on SIP/voipjet-00708330 [Mar 14 19:03:49] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK4d6faf5e;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as35bd9002 To: Call-ID: 1ff878460fa6757e752e68a0704c27c9@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 19:03:49] --- (10 headers 0 lines) --- [Mar 14 19:03:51] <--- 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=z9hG4bK4d6faf5e;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as35bd9002 To: ;tag=as44528ce3 Call-ID: 1ff878460fa6757e752e68a0704c27c9@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 16382 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 19:03:51] --- (11 headers 10 lines) --- [Mar 14 19:03:51] Found RTP audio format 0 [Mar 14 19:03:51] Found RTP audio format 101 [Mar 14 19:03:51] Peer audio RTP is at port 69.16.233.35:16382 [Mar 14 19:03:51] Found description format PCMU for ID 0 [Mar 14 19:03:51] Found description format telephone-event for ID 101 [Mar 14 19:03:51] Got unsupported a:fmtp in SDP offer [Mar 14 19:03:51] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:03:51] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:03:51] Peer audio RTP is at port 69.16.233.35:16382 [Mar 14 19:03:51] -- Call on SIP/voipjet-ae00b8c0 left from hold [Mar 14 19:03:51] -- Stopped music on hold on SIP/voipjet-00708330 [Mar 14 19:03:51] -- SIP/voipjet-ae00b8c0 is making progress passing it to SIP/voipjet-00708330 [Mar 14 19:03:54] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK4d6faf5e;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as35bd9002 To: ;tag=as44528ce3 Call-ID: 1ff878460fa6757e752e68a0704c27c9@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 16382 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 19:03:54] --- (11 headers 10 lines) --- [Mar 14 19:03:54] Found RTP audio format 0 [Mar 14 19:03:54] Found RTP audio format 101 [Mar 14 19:03:54] Peer audio RTP is at port 69.16.233.35:16382 [Mar 14 19:03:54] Found description format PCMU for ID 0 [Mar 14 19:03:54] Found description format telephone-event for ID 101 [Mar 14 19:03:54] Got unsupported a:fmtp in SDP offer [Mar 14 19:03:54] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:03:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:03:54] Peer audio RTP is at port 69.16.233.35:16382 [Mar 14 19:03:54] list_route: hop: [Mar 14 19:03:54] set_destination: Parsing for address/port to send to [Mar 14 19:03:54] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:03:54] 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=z9hG4bK76becf53;rport From: "3024573281" ;tag=as35bd9002 To: ;tag=as44528ce3 Contact: Call-ID: 1ff878460fa6757e752e68a0704c27c9@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:03:54] -- Call on SIP/voipjet-ae00b8c0 left from hold [Mar 14 19:03:54] -- SIP/voipjet-ae00b8c0 answered SIP/voipjet-00708330 [Mar 14 19:04:05] <--- 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=z9hG4bK424b1b1e;rport From: ;tag=as619bae0b To: "6025161069" ;tag=as70c79f3a Call-ID: 26e9cc1668f3e45520324f28541304b2@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 19:04:05] --- (9 headers 0 lines) --- [Mar 14 19:04:05] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 19:04:05] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK424b1b1e;received=69.16.233.35;rport=5060 From: ;tag=as619bae0b To: "6025161069" ;tag=as70c79f3a Call-ID: 26e9cc1668f3e45520324f28541304b2@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 19:04:05] Scheduling destruction of SIP dialog '1ff878460fa6757e752e68a0704c27c9@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 19:04:05] set_destination: Parsing for address/port to send to [Mar 14 19:04:05] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:04:05] 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=z9hG4bK14087804;rport From: "3024573281" ;tag=as35bd9002 To: ;tag=as44528ce3 Call-ID: 1ff878460fa6757e752e68a0704c27c9@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:04:05] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-00708330' [Mar 14 19:04:05] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-00708330", "CDR(userfield)=press1transfer 20070314190349-1173913406.43.ulaw") in new stack [Mar 14 19:04:05] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-00708330", "CDR(userfield2)=5083282553") in new stack [Mar 14 19:04:05] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 19:04:05] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-00708330", "CDR(accountcode)=479") in new stack [Mar 14 19:04:05] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-00708330", "5083282553") in new stack [Mar 14 19:04:05] == End MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:04:05] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK14087804;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as35bd9002 To: ;tag=as44528ce3 Call-ID: 1ff878460fa6757e752e68a0704c27c9@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 19:04:05] --- (11 headers 0 lines) --- [Mar 14 19:04:05] Really destroying SIP dialog '1ff878460fa6757e752e68a0704c27c9@167.206.178.7' Method: INVITE [Mar 14 19:04:05] RTP-stats [Mar 14 19:04:05] * Our Receiver: [Mar 14 19:04:05] SSRC: 1305674410 [Mar 14 19:04:05] Received packets: 688 [Mar 14 19:04:05] Lost packets: 0 [Mar 14 19:04:05] Jitter: 0.0001 [Mar 14 19:04:05] Transit: -0.0003 [Mar 14 19:04:05] RR-count: 0 [Mar 14 19:04:05] * Our Sender: [Mar 14 19:04:05] SSRC: 1061834332 [Mar 14 19:04:05] Sent packets: 533 [Mar 14 19:04:05] Lost packets: 0 [Mar 14 19:04:05] Jitter: 0 [Mar 14 19:04:05] SR-count: 2 [Mar 14 19:04:05] RTT: 0.000000 [Mar 14 19:04:05] Really destroying SIP dialog '26e9cc1668f3e45520324f28541304b2@167.206.178.7' Method: BYE [Mar 14 19:04:05] RTP-stats [Mar 14 19:04:05] * Our Receiver: [Mar 14 19:04:05] SSRC: 1516613930 [Mar 14 19:04:05] Received packets: 1617 [Mar 14 19:04:05] Lost packets: 3 [Mar 14 19:04:05] Jitter: 0.0002 [Mar 14 19:04:05] Transit: -0.0001 [Mar 14 19:04:05] RR-count: 1 [Mar 14 19:04:05] * Our Sender: [Mar 14 19:04:05] SSRC: 1051805076 [Mar 14 19:04:05] Sent packets: 1012 [Mar 14 19:04:05] Lost packets: 0 [Mar 14 19:04:05] Jitter: 0 [Mar 14 19:04:05] SR-count: 5 [Mar 14 19:04:05] RTT: 0.000000