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 18:45:40] 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=z9hG4bK7e628bfd;rport From: "asterisk" ;tag=as1f037f55 To: Contact: Call-ID: 4b70a9e6323cbfc56d69f03a3a2d1e03@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:45:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:45:40] <--- 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=z9hG4bK7e628bfd;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as1f037f55 To: ;tag=as4ba9ccbe Call-ID: 4b70a9e6323cbfc56d69f03a3a2d1e03@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 18:45:40] --- (10 headers 0 lines) --- [Mar 14 18:45:40] Really destroying SIP dialog '4b70a9e6323cbfc56d69f03a3a2d1e03@167.206.178.7' Method: OPTIONS [Mar 14 18:45:41] 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=z9hG4bK417fe7de;rport From: "asterisk" ;tag=as0bbd3478 To: Contact: Call-ID: 64cfd9a420829218249531113d335399@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:45:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:45:41] <--- 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=z9hG4bK417fe7de;rport From: "asterisk" ;tag=as0bbd3478 To: Call-ID: 64cfd9a420829218249531113d335399@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 18:45:41] --- (7 headers 0 lines) --- [Mar 14 18:45:41] Really destroying SIP dialog '64cfd9a420829218249531113d335399@167.206.178.7' Method: OPTIONS [Mar 14 18:45:42] 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=z9hG4bK4087cf5a;rport From: "asterisk" ;tag=as066dc41d To: Contact: Call-ID: 5b8f819751c611814e0171886df03e83@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:45:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:45:42] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK4087cf5a;rport=5060 From: "asterisk" ;tag=as066dc41d To: ;tag=0-tdb3221212128 Call-ID: 5b8f819751c611814e0171886df03e83@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 18:45:42] --- (9 headers 0 lines) --- [Mar 14 18:45:42] Really destroying SIP dialog '5b8f819751c611814e0171886df03e83@167.206.178.7' Method: OPTIONS [Mar 14 18:46:37] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:46:37] Found [Mar 14 18:46:40] 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=z9hG4bK73413fb9;rport From: "asterisk" ;tag=as39a7c3c8 To: Contact: Call-ID: 7006545b5bedda051db6be7a15661fca@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:46:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:46:40] <--- 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=z9hG4bK73413fb9;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as39a7c3c8 To: ;tag=as0443ffa8 Call-ID: 7006545b5bedda051db6be7a15661fca@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 18:46:40] --- (10 headers 0 lines) --- [Mar 14 18:46:40] Really destroying SIP dialog '7006545b5bedda051db6be7a15661fca@167.206.178.7' Method: OPTIONS [Mar 14 18:46:41] 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=z9hG4bK4b06791e;rport From: "asterisk" ;tag=as46d15b09 To: Contact: Call-ID: 00ba14397bf17571017cad9f714da809@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:46:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:46:41] <--- 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=z9hG4bK4b06791e;rport From: "asterisk" ;tag=as46d15b09 To: Call-ID: 00ba14397bf17571017cad9f714da809@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 18:46:41] --- (7 headers 0 lines) --- [Mar 14 18:46:41] Really destroying SIP dialog '00ba14397bf17571017cad9f714da809@167.206.178.7' Method: OPTIONS [Mar 14 18:46:42] 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=z9hG4bK57f0d96c;rport From: "asterisk" ;tag=as6902b23b To: Contact: Call-ID: 549c28cb2b154fb737cad4f356ce5448@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:46:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:46:42] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK57f0d96c;rport=5060 From: "asterisk" ;tag=as6902b23b To: ;tag=0-tdb3221212128 Call-ID: 549c28cb2b154fb737cad4f356ce5448@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 18:46:42] --- (9 headers 0 lines) --- [Mar 14 18:46:42] Really destroying SIP dialog '549c28cb2b154fb737cad4f356ce5448@167.206.178.7' Method: OPTIONS [Mar 14 18:46:56] RTP-stats [Mar 14 18:46:56] * Our Receiver: [Mar 14 18:46:56] SSRC: 0 [Mar 14 18:46:56] Received packets: 0 [Mar 14 18:46:56] Lost packets: 0 [Mar 14 18:46:56] Jitter: 0.0000 [Mar 14 18:46:56] Transit: 0.0000 [Mar 14 18:46:56] RR-count: 0 [Mar 14 18:46:56] * Our Sender: [Mar 14 18:46:56] SSRC: 276192733 [Mar 14 18:46:56] Sent packets: 0 [Mar 14 18:46:56] Lost packets: 0 [Mar 14 18:46:56] Jitter: 0 [Mar 14 18:46:56] SR-count: 0 [Mar 14 18:46:56] RTT: 0.000000 [Mar 14 18:46:56] Audio is at 167.206.178.7 port 17058 [Mar 14 18:46:56] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:46:56] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:46:56] 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=z9hG4bK5d10bb7e;rport From: "6025161069" ;tag=as64d426fe To: Contact: Call-ID: 17a7841c7657343e5b270de27ba99ffa@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:46:56 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 17058 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 18:46:56] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d10bb7e;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as64d426fe To: Call-ID: 17a7841c7657343e5b270de27ba99ffa@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 18:46:56] --- (10 headers 0 lines) --- [Mar 14 18:47:02] <--- 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=z9hG4bK5d10bb7e;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as64d426fe To: ;tag=as49304790 Call-ID: 17a7841c7657343e5b270de27ba99ffa@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 14462 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 18:47:02] --- (11 headers 10 lines) --- [Mar 14 18:47:02] Found RTP audio format 0 [Mar 14 18:47:02] Found RTP audio format 101 [Mar 14 18:47:02] Peer audio RTP is at port 69.16.233.35:14462 [Mar 14 18:47:02] Found description format PCMU for ID 0 [Mar 14 18:47:02] Found description format telephone-event for ID 101 [Mar 14 18:47:02] Got unsupported a:fmtp in SDP offer [Mar 14 18:47:02] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:47:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:47:02] Peer audio RTP is at port 69.16.233.35:14462 [Mar 14 18:47:08] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK5d10bb7e;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as64d426fe To: ;tag=as49304790 Call-ID: 17a7841c7657343e5b270de27ba99ffa@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 14462 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 18:47:08] --- (11 headers 10 lines) --- [Mar 14 18:47:08] Found RTP audio format 0 [Mar 14 18:47:08] Found RTP audio format 101 [Mar 14 18:47:08] Peer audio RTP is at port 69.16.233.35:14462 [Mar 14 18:47:08] Found description format PCMU for ID 0 [Mar 14 18:47:08] Found description format telephone-event for ID 101 [Mar 14 18:47:08] Got unsupported a:fmtp in SDP offer [Mar 14 18:47:08] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:47:08] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:47:08] Peer audio RTP is at port 69.16.233.35:14462 [Mar 14 18:47:08] list_route: hop: [Mar 14 18:47:08] set_destination: Parsing for address/port to send to [Mar 14 18:47:08] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:47:08] 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=z9hG4bK72fc0fca;rport From: "6025161069" ;tag=as64d426fe To: ;tag=as49304790 Contact: Call-ID: 17a7841c7657343e5b270de27ba99ffa@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:47:08] > Channel SIP/voipjet-0070afe0 was answered. [Mar 14 18:47:08] -- Executing [s@liveonly:1] Answer("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:47:08] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-0070afe0", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:476 CALLERID: ") in new stack [Mar 14 18:47:08] -- Executing [s@liveonly:3] Wait("SIP/voipjet-0070afe0", "1") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:4] Set("SIP/voipjet-0070afe0", "soundfile=DEBUG-20070314184709-1173912416.36.wav") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-0070afe0", "DEBUG-20070314184709-1173912416.36.wav") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:6] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=476") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:7] Set("SIP/voipjet-0070afe0", "CDR(userfield)= 5083282553") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-0070afe0", "machinedetect") in new stack [Mar 14 18:47:09] -- Executing [s@liveonly:9] AMD("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:47:09] -- AMD: SIP/voipjet-0070afe0 6025161069 (null) (Fmt: 4) [Mar 14 18:47:09] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 18:47:11] -- AMD: Word detected. iWordsCount:1 [Mar 14 18:47:11] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 18:47:12] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 18:47:12] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-0070afe0", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 18:47:12] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-0070afe0", "0?liveonly-amd|s|1") in new stack [Mar 14 18:47:12] -- Executing [s@liveonly:12] Goto("SIP/voipjet-0070afe0", "liveonly-live|s|1") in new stack [Mar 14 18:47:12] -- Goto (liveonly-live,s,1) [Mar 14 18:47:12] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-0070afe0", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 18:47:12] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 18:47:16] == CDR updated on SIP/voipjet-0070afe0 [Mar 14 18:47:16] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-0070afe0", "Press 1 transfer") in new stack [Mar 14 18:47:16] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-0070afe0", "liveonly-transfer|s|1") in new stack [Mar 14 18:47:16] -- Goto (liveonly-transfer,s,1) [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "soundfile=20070314184716-1173912416.36.ulaw") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-0070afe0", "20070314184716-1173912416.36.ulaw|b") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CALLERID(num)=3024573281") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-0070afe0", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-0070afe0", "CDR(userfield5)=2007-03-14 18:47:16") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-0070afe0", "0?20") in new stack [Mar 14 18:47:16] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-0070afe0", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 18:47:16] RTP-stats [Mar 14 18:47:16] * Our Receiver: [Mar 14 18:47:16] SSRC: 0 [Mar 14 18:47:16] Received packets: 0 [Mar 14 18:47:16] Lost packets: 0 [Mar 14 18:47:16] Jitter: 0.0000 [Mar 14 18:47:16] Transit: 0.0000 [Mar 14 18:47:16] RR-count: 0 [Mar 14 18:47:16] * Our Sender: [Mar 14 18:47:16] SSRC: 1892302993 [Mar 14 18:47:16] Sent packets: 0 [Mar 14 18:47:16] Lost packets: 0 [Mar 14 18:47:16] Jitter: 0 [Mar 14 18:47:16] SR-count: 0 [Mar 14 18:47:16] RTT: 0.000000 [Mar 14 18:47:16] == Begin MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:47:16] Audio is at 167.206.178.7 port 17048 [Mar 14 18:47:16] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:47:16] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:47:16] 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=z9hG4bK6606562d;rport From: "3024573281" ;tag=as4fb437bd To: Contact: Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:47:16 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 17048 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 18:47:16] -- Called 13024573281@voipjet [Mar 14 18:47:16] -- Started music on hold, class 'trans3', on SIP/voipjet-0070afe0 [Mar 14 18:47:16] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6606562d;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4fb437bd To: Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@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 18:47:16] --- (10 headers 0 lines) --- [Mar 14 18:47:18] <--- 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=z9hG4bK6606562d;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4fb437bd To: ;tag=as01946992 Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@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 18742 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 18:47:18] --- (11 headers 10 lines) --- [Mar 14 18:47:18] Found RTP audio format 0 [Mar 14 18:47:18] Found RTP audio format 101 [Mar 14 18:47:18] Peer audio RTP is at port 69.16.233.35:18742 [Mar 14 18:47:18] Found description format PCMU for ID 0 [Mar 14 18:47:18] Found description format telephone-event for ID 101 [Mar 14 18:47:18] Got unsupported a:fmtp in SDP offer [Mar 14 18:47:18] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:47:18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:47:18] Peer audio RTP is at port 69.16.233.35:18742 [Mar 14 18:47:18] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:47:18] -- Stopped music on hold on SIP/voipjet-0070afe0 [Mar 14 18:47:18] -- SIP/voipjet-006c0ff0 is making progress passing it to SIP/voipjet-0070afe0 [Mar 14 18:47:21] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6606562d;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4fb437bd To: ;tag=as01946992 Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@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 18742 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 18:47:21] --- (11 headers 10 lines) --- [Mar 14 18:47:21] Found RTP audio format 0 [Mar 14 18:47:21] Found RTP audio format 101 [Mar 14 18:47:21] Peer audio RTP is at port 69.16.233.35:18742 [Mar 14 18:47:21] Found description format PCMU for ID 0 [Mar 14 18:47:21] Found description format telephone-event for ID 101 [Mar 14 18:47:21] Got unsupported a:fmtp in SDP offer [Mar 14 18:47:21] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:47:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:47:21] Peer audio RTP is at port 69.16.233.35:18742 [Mar 14 18:47:21] list_route: hop: [Mar 14 18:47:21] set_destination: Parsing for address/port to send to [Mar 14 18:47:21] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:47:21] 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=z9hG4bK7b7349c7;rport From: "3024573281" ;tag=as4fb437bd To: ;tag=as01946992 Contact: Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:47:21] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:47:21] -- SIP/voipjet-006c0ff0 answered SIP/voipjet-0070afe0 [Mar 14 18:47:33] <--- 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=z9hG4bK55cb651f;rport From: ;tag=as49304790 To: "6025161069" ;tag=as64d426fe Call-ID: 17a7841c7657343e5b270de27ba99ffa@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 18:47:33] --- (9 headers 0 lines) --- [Mar 14 18:47:33] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 18:47:33] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK55cb651f;received=69.16.233.35;rport=5060 From: ;tag=as49304790 To: "6025161069" ;tag=as64d426fe Call-ID: 17a7841c7657343e5b270de27ba99ffa@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 18:47:33] Scheduling destruction of SIP dialog '6835f9557d9c191f3ec4e4e220e3350f@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 18:47:33] set_destination: Parsing for address/port to send to [Mar 14 18:47:33] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:47:33] 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=z9hG4bK52f789a2;rport From: "3024573281" ;tag=as4fb437bd To: ;tag=as01946992 Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:47:33] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-0070afe0' [Mar 14 18:47:33] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "CDR(userfield)=press1transfer 20070314184716-1173912416.36.ulaw") in new stack [Mar 14 18:47:33] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-0070afe0", "CDR(userfield2)=5083282553") in new stack [Mar 14 18:47:33] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 18:47:33] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=476") in new stack [Mar 14 18:47:33] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-0070afe0", "5083282553") in new stack [Mar 14 18:47:33] == End MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:47:33] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK52f789a2;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4fb437bd To: ;tag=as01946992 Call-ID: 6835f9557d9c191f3ec4e4e220e3350f@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 18:47:33] --- (11 headers 0 lines) --- [Mar 14 18:47:33] Really destroying SIP dialog '6835f9557d9c191f3ec4e4e220e3350f@167.206.178.7' Method: INVITE [Mar 14 18:47:33] RTP-stats [Mar 14 18:47:33] * Our Receiver: [Mar 14 18:47:33] SSRC: 313673140 [Mar 14 18:47:33] Received packets: 769 [Mar 14 18:47:33] Lost packets: 0 [Mar 14 18:47:33] Jitter: 0.0002 [Mar 14 18:47:33] Transit: 0.0000 [Mar 14 18:47:33] RR-count: 0 [Mar 14 18:47:33] * Our Sender: [Mar 14 18:47:33] SSRC: 1034188353 [Mar 14 18:47:33] Sent packets: 602 [Mar 14 18:47:33] Lost packets: 0 [Mar 14 18:47:33] Jitter: 0 [Mar 14 18:47:33] SR-count: 3 [Mar 14 18:47:33] RTT: 0.000000 [Mar 14 18:47:33] Really destroying SIP dialog '17a7841c7657343e5b270de27ba99ffa@167.206.178.7' Method: BYE [Mar 14 18:47:33] RTP-stats [Mar 14 18:47:33] * Our Receiver: [Mar 14 18:47:33] SSRC: 1105255764 [Mar 14 18:47:33] Received packets: 1501 [Mar 14 18:47:33] Lost packets: 16 [Mar 14 18:47:33] Jitter: 0.0003 [Mar 14 18:47:33] Transit: 0.0009 [Mar 14 18:47:33] RR-count: 1 [Mar 14 18:47:33] * Our Sender: [Mar 14 18:47:33] SSRC: 626983927 [Mar 14 18:47:33] Sent packets: 881 [Mar 14 18:47:33] Lost packets: 0 [Mar 14 18:47:33] Jitter: 0 [Mar 14 18:47:33] SR-count: 5 [Mar 14 18:47:33] RTT: 0.000000