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 19:05:26] RTP-stats [Mar 14 19:05:26] * Our Receiver: [Mar 14 19:05:26] SSRC: 0 [Mar 14 19:05:26] Received packets: 0 [Mar 14 19:05:26] Lost packets: 0 [Mar 14 19:05:26] Jitter: 0.0000 [Mar 14 19:05:26] Transit: 0.0000 [Mar 14 19:05:26] RR-count: 0 [Mar 14 19:05:26] * Our Sender: [Mar 14 19:05:26] SSRC: 317420366 [Mar 14 19:05:26] Sent packets: 0 [Mar 14 19:05:26] Lost packets: 0 [Mar 14 19:05:26] Jitter: 0 [Mar 14 19:05:26] SR-count: 0 [Mar 14 19:05:26] RTT: 0.000000 [Mar 14 19:05:26] Audio is at 167.206.178.7 port 19680 [Mar 14 19:05:26] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:05:26] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:05:26] 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=z9hG4bK3a9d89a0;rport From: "6025161069" ;tag=as7db13b26 To: Contact: Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:05:26 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 19680 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 19:05:26] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK3a9d89a0;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7db13b26 To: Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@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 19:05:26] --- (10 headers 0 lines) --- [Mar 14 19:05:31] <--- 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=z9hG4bK3a9d89a0;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7db13b26 To: ;tag=as2f3b86cf Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@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 14118 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 19:05:31] --- (11 headers 10 lines) --- [Mar 14 19:05:31] Found RTP audio format 0 [Mar 14 19:05:31] Found RTP audio format 101 [Mar 14 19:05:31] Peer audio RTP is at port 69.16.233.35:14118 [Mar 14 19:05:31] Found description format PCMU for ID 0 [Mar 14 19:05:31] Found description format telephone-event for ID 101 [Mar 14 19:05:31] Got unsupported a:fmtp in SDP offer [Mar 14 19:05:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:05:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:05:31] Peer audio RTP is at port 69.16.233.35:14118 [Mar 14 19:05:32] 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=z9hG4bK5dc76527;rport From: "asterisk" ;tag=as7b9cbf2e To: Contact: Call-ID: 5f753c021b469c0f7dfd53662d0f4429@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:05:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:05:32] <--- 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=z9hG4bK5dc76527;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as7b9cbf2e To: ;tag=as69e3a665 Call-ID: 5f753c021b469c0f7dfd53662d0f4429@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 19:05:32] --- (10 headers 0 lines) --- [Mar 14 19:05:32] Really destroying SIP dialog '5f753c021b469c0f7dfd53662d0f4429@167.206.178.7' Method: OPTIONS [Mar 14 19:05:33] 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=z9hG4bK478a7cd5;rport From: "asterisk" ;tag=as05931f1d To: Contact: Call-ID: 1d0b9ac8173724fc09633f1175b6e480@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:05:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:05:33] <--- 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=z9hG4bK478a7cd5;rport From: "asterisk" ;tag=as05931f1d To: Call-ID: 1d0b9ac8173724fc09633f1175b6e480@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 19:05:33] --- (7 headers 0 lines) --- [Mar 14 19:05:33] Really destroying SIP dialog '1d0b9ac8173724fc09633f1175b6e480@167.206.178.7' Method: OPTIONS [Mar 14 19:05:33] 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=z9hG4bK7617fce2;rport From: "asterisk" ;tag=as35acdc28 To: Contact: Call-ID: 26ad00413b8b7762214f84bf5e72e423@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:05:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:05:33] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK7617fce2;rport=5060 From: "asterisk" ;tag=as35acdc28 To: ;tag=0-tdb3221212128 Call-ID: 26ad00413b8b7762214f84bf5e72e423@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 19:05:33] --- (9 headers 0 lines) --- [Mar 14 19:05:33] Really destroying SIP dialog '26ad00413b8b7762214f84bf5e72e423@167.206.178.7' Method: OPTIONS [Mar 14 19:05:39] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK3a9d89a0;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7db13b26 To: ;tag=as2f3b86cf Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@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 14118 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 19:05:39] --- (11 headers 10 lines) --- [Mar 14 19:05:39] Found RTP audio format 0 [Mar 14 19:05:39] Found RTP audio format 101 [Mar 14 19:05:39] Peer audio RTP is at port 69.16.233.35:14118 [Mar 14 19:05:39] Found description format PCMU for ID 0 [Mar 14 19:05:39] Found description format telephone-event for ID 101 [Mar 14 19:05:39] Got unsupported a:fmtp in SDP offer [Mar 14 19:05:39] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:05:39] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:05:39] Peer audio RTP is at port 69.16.233.35:14118 [Mar 14 19:05:39] list_route: hop: [Mar 14 19:05:39] set_destination: Parsing for address/port to send to [Mar 14 19:05:39] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:05:39] 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=z9hG4bK1b9e9592;rport From: "6025161069" ;tag=as7db13b26 To: ;tag=as2f3b86cf Contact: Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:05:39] > Channel SIP/voipjet-00708330 was answered. [Mar 14 19:05:39] -- Executing [s@liveonly:1] Answer("SIP/voipjet-00708330", "") in new stack [Mar 14 19:05:39] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-00708330", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:480 CALLERID: ") in new stack [Mar 14 19:05:39] -- Executing [s@liveonly:3] Wait("SIP/voipjet-00708330", "1") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:4] Set("SIP/voipjet-00708330", "soundfile=DEBUG-20070314190540-1173913526.45.wav") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-00708330", "DEBUG-20070314190540-1173913526.45.wav") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:6] Set("SIP/voipjet-00708330", "CDR(accountcode)=480") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:7] Set("SIP/voipjet-00708330", "CDR(userfield)= 5083282553") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-00708330", "machinedetect") in new stack [Mar 14 19:05:40] -- Executing [s@liveonly:9] AMD("SIP/voipjet-00708330", "") in new stack [Mar 14 19:05:40] -- AMD: SIP/voipjet-00708330 6025161069 (null) (Fmt: 4) [Mar 14 19:05:40] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 19:05:42] -- AMD: Word detected. iWordsCount:1 [Mar 14 19:05:42] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 19:05:43] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 19:05:43] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-00708330", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 19:05:43] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-00708330", "0?liveonly-amd|s|1") in new stack [Mar 14 19:05:43] -- Executing [s@liveonly:12] Goto("SIP/voipjet-00708330", "liveonly-live|s|1") in new stack [Mar 14 19:05:43] -- Goto (liveonly-live,s,1) [Mar 14 19:05:43] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-00708330", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 19:05:43] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 19:05:47] == CDR updated on SIP/voipjet-00708330 [Mar 14 19:05:47] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-00708330", "Press 1 transfer") in new stack [Mar 14 19:05:47] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-00708330", "liveonly-transfer|s|1") in new stack [Mar 14 19:05:47] -- Goto (liveonly-transfer,s,1) [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-00708330", "soundfile=20070314190547-1173913526.45.ulaw") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-00708330", "20070314190547-1173913526.45.ulaw|b") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CALLERID(num)=3024573281") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-00708330", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-00708330", "CDR(userfield5)=2007-03-14 19:05:47") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-00708330", "0?20") in new stack [Mar 14 19:05:47] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-00708330", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 19:05:47] RTP-stats [Mar 14 19:05:47] * Our Receiver: [Mar 14 19:05:47] == Begin MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:05:47] SSRC: 0 [Mar 14 19:05:47] Received packets: 0 [Mar 14 19:05:47] Lost packets: 0 [Mar 14 19:05:47] Jitter: 0.0000 [Mar 14 19:05:47] Transit: 0.0000 [Mar 14 19:05:47] RR-count: 0 [Mar 14 19:05:47] * Our Sender: [Mar 14 19:05:47] SSRC: 1288812395 [Mar 14 19:05:47] Sent packets: 0 [Mar 14 19:05:47] Lost packets: 0 [Mar 14 19:05:47] Jitter: 0 [Mar 14 19:05:47] SR-count: 0 [Mar 14 19:05:47] RTT: 0.000000 [Mar 14 19:05:47] Audio is at 167.206.178.7 port 19028 [Mar 14 19:05:47] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:05:47] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:05:47] 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=z9hG4bK194e1ebe;rport From: "3024573281" ;tag=as3fb58ece To: Contact: Call-ID: 41760ba824217f3a6dc33302432a59da@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:05:47 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 19028 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 19:05:47] -- Called 13024573281@voipjet [Mar 14 19:05:47] -- Started music on hold, class 'trans3', on SIP/voipjet-00708330 [Mar 14 19:05:47] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK194e1ebe;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3fb58ece To: Call-ID: 41760ba824217f3a6dc33302432a59da@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 19:05:47] --- (10 headers 0 lines) --- [Mar 14 19:05:49] <--- 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=z9hG4bK194e1ebe;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3fb58ece To: ;tag=as76eac0d4 Call-ID: 41760ba824217f3a6dc33302432a59da@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 12784 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 19:05:49] --- (11 headers 10 lines) --- [Mar 14 19:05:49] Found RTP audio format 0 [Mar 14 19:05:49] Found RTP audio format 101 [Mar 14 19:05:49] Peer audio RTP is at port 69.16.233.35:12784 [Mar 14 19:05:49] Found description format PCMU for ID 0 [Mar 14 19:05:49] Found description format telephone-event for ID 101 [Mar 14 19:05:49] Got unsupported a:fmtp in SDP offer [Mar 14 19:05:49] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:05:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:05:49] Peer audio RTP is at port 69.16.233.35:12784 [Mar 14 19:05:49] -- Call on SIP/voipjet-00726fa0 left from hold [Mar 14 19:05:49] -- Stopped music on hold on SIP/voipjet-00708330 [Mar 14 19:05:49] -- SIP/voipjet-00726fa0 is making progress passing it to SIP/voipjet-00708330 [Mar 14 19:05:52] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK194e1ebe;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3fb58ece To: ;tag=as76eac0d4 Call-ID: 41760ba824217f3a6dc33302432a59da@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 12784 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 19:05:52] --- (11 headers 10 lines) --- [Mar 14 19:05:52] Found RTP audio format 0 [Mar 14 19:05:52] Found RTP audio format 101 [Mar 14 19:05:52] Peer audio RTP is at port 69.16.233.35:12784 [Mar 14 19:05:52] Found description format PCMU for ID 0 [Mar 14 19:05:52] Found description format telephone-event for ID 101 [Mar 14 19:05:52] Got unsupported a:fmtp in SDP offer [Mar 14 19:05:52] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:05:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:05:52] Peer audio RTP is at port 69.16.233.35:12784 [Mar 14 19:05:52] list_route: hop: [Mar 14 19:05:52] set_destination: Parsing for address/port to send to [Mar 14 19:05:52] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:05:52] 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=z9hG4bK15627829;rport From: "3024573281" ;tag=as3fb58ece To: ;tag=as76eac0d4 Contact: Call-ID: 41760ba824217f3a6dc33302432a59da@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:05:52] -- Call on SIP/voipjet-00726fa0 left from hold [Mar 14 19:05:52] -- SIP/voipjet-00726fa0 answered SIP/voipjet-00708330 [Mar 14 19:06:02] <--- 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=z9hG4bK5af0b47e;rport From: ;tag=as2f3b86cf To: "6025161069" ;tag=as7db13b26 Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 19:06:02] --- (9 headers 0 lines) --- [Mar 14 19:06:02] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 19:06:02] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK5af0b47e;received=69.16.233.35;rport=5060 From: ;tag=as2f3b86cf To: "6025161069" ;tag=as7db13b26 Call-ID: 3d3c7e2e09e81a213a58217b5a5099c1@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 19:06:02] Scheduling destruction of SIP dialog '41760ba824217f3a6dc33302432a59da@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 19:06:02] set_destination: Parsing for address/port to send to [Mar 14 19:06:02] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:06:02] 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=z9hG4bK12ab5dec;rport From: "3024573281" ;tag=as3fb58ece To: ;tag=as76eac0d4 Call-ID: 41760ba824217f3a6dc33302432a59da@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:06:02] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-00708330' [Mar 14 19:06:02] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-00708330", "CDR(userfield)=press1transfer 20070314190547-1173913526.45.ulaw") in new stack [Mar 14 19:06:02] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-00708330", "CDR(userfield2)=5083282553") in new stack [Mar 14 19:06:02] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 19:06:02] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-00708330", "CDR(accountcode)=480") in new stack [Mar 14 19:06:02] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-00708330", "5083282553") in new stack [Mar 14 19:06:02] == End MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:06:02] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK12ab5dec;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3fb58ece To: ;tag=as76eac0d4 Call-ID: 41760ba824217f3a6dc33302432a59da@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 19:06:02] --- (11 headers 0 lines) --- [Mar 14 19:06:02] Really destroying SIP dialog '41760ba824217f3a6dc33302432a59da@167.206.178.7' Method: INVITE [Mar 14 19:06:02] RTP-stats [Mar 14 19:06:02] * Our Receiver: [Mar 14 19:06:02] SSRC: 1771494905 [Mar 14 19:06:02] Received packets: 643 [Mar 14 19:06:02] Lost packets: 0 [Mar 14 19:06:02] Jitter: 0.0001 [Mar 14 19:06:02] Transit: -0.0000 [Mar 14 19:06:02] RR-count: 0 [Mar 14 19:06:02] * Our Sender: [Mar 14 19:06:02] SSRC: 233188524 [Mar 14 19:06:02] Sent packets: 469 [Mar 14 19:06:02] Lost packets: 0 [Mar 14 19:06:02] Jitter: 0 [Mar 14 19:06:02] SR-count: 2 [Mar 14 19:06:02] RTT: 0.000000 [Mar 14 19:06:02] Really destroying SIP dialog '3d3c7e2e09e81a213a58217b5a5099c1@167.206.178.7' Method: BYE [Mar 14 19:06:02] RTP-stats [Mar 14 19:06:02] * Our Receiver: [Mar 14 19:06:02] SSRC: 657261732 [Mar 14 19:06:02] Received packets: 1503 [Mar 14 19:06:02] Lost packets: 0 [Mar 14 19:06:02] Jitter: 0.0002 [Mar 14 19:06:02] Transit: -0.0002 [Mar 14 19:06:02] RR-count: 2 [Mar 14 19:06:02] * Our Sender: [Mar 14 19:06:02] SSRC: 1133066289 [Mar 14 19:06:02] Sent packets: 768 [Mar 14 19:06:02] Lost packets: 0 [Mar 14 19:06:02] Jitter: 0 [Mar 14 19:06:02] SR-count: 4 [Mar 14 19:06:02] RTT: 0.000000