[Mar 14 18:59:57] RTP-stats [Mar 14 18:59:57] * Our Receiver: [Mar 14 18:59:57] SSRC: 0 [Mar 14 18:59:57] Received packets: 0 [Mar 14 18:59:57] Lost packets: 0 [Mar 14 18:59:57] Jitter: 0.0000 [Mar 14 18:59:57] Transit: 0.0000 [Mar 14 18:59:57] RR-count: 0 [Mar 14 18:59:57] * Our Sender: [Mar 14 18:59:57] SSRC: 124647459 [Mar 14 18:59:57] Sent packets: 0 [Mar 14 18:59:57] Lost packets: 0 [Mar 14 18:59:57] Jitter: 0 [Mar 14 18:59:57] SR-count: 0 [Mar 14 18:59:57] RTT: 0.000000 [Mar 14 18:59:57] Audio is at 167.206.178.7 port 18402 [Mar 14 18:59:57] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:59:57] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:59:57] 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=z9hG4bK1e22ee5d;rport From: "6025161069" ;tag=as2a076c81 To: Contact: Call-ID: 40c041d838afa4f41165ad9421b604cc@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:59:57 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 18402 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 18:59:57] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK1e22ee5d;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as2a076c81 To: Call-ID: 40c041d838afa4f41165ad9421b604cc@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 18:59:57] --- (10 headers 0 lines) --- [Mar 14 19:00:03] <--- 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=z9hG4bK1e22ee5d;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as2a076c81 To: ;tag=as2747d21c Call-ID: 40c041d838afa4f41165ad9421b604cc@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 11722 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 19:00:03] --- (11 headers 10 lines) --- [Mar 14 19:00:03] Found RTP audio format 0 [Mar 14 19:00:03] Found RTP audio format 101 [Mar 14 19:00:03] Peer audio RTP is at port 69.16.233.35:11722 [Mar 14 19:00:03] Found description format PCMU for ID 0 [Mar 14 19:00:03] Found description format telephone-event for ID 101 [Mar 14 19:00:03] Got unsupported a:fmtp in SDP offer [Mar 14 19:00:03] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:00:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:00:03] Peer audio RTP is at port 69.16.233.35:11722 [Mar 14 19:00:05] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK1e22ee5d;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as2a076c81 To: ;tag=as2747d21c Call-ID: 40c041d838afa4f41165ad9421b604cc@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 11722 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 19:00:05] --- (11 headers 10 lines) --- [Mar 14 19:00:05] Found RTP audio format 0 [Mar 14 19:00:05] Found RTP audio format 101 [Mar 14 19:00:05] Peer audio RTP is at port 69.16.233.35:11722 [Mar 14 19:00:05] Found description format PCMU for ID 0 [Mar 14 19:00:05] Found description format telephone-event for ID 101 [Mar 14 19:00:05] Got unsupported a:fmtp in SDP offer [Mar 14 19:00:05] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:00:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:00:05] Peer audio RTP is at port 69.16.233.35:11722 [Mar 14 19:00:05] list_route: hop: [Mar 14 19:00:05] set_destination: Parsing for address/port to send to [Mar 14 19:00:05] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:00:05] 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=z9hG4bK085c873e;rport From: "6025161069" ;tag=as2a076c81 To: ;tag=as2747d21c Contact: Call-ID: 40c041d838afa4f41165ad9421b604cc@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:00:05] > Channel SIP/voipjet-00708330 was answered. [Mar 14 19:00:05] -- Executing [s@liveonly:1] Answer("SIP/voipjet-00708330", "") in new stack [Mar 14 19:00:05] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-00708330", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:477 CALLERID: ") in new stack [Mar 14 19:00:05] -- Executing [s@liveonly:3] Wait("SIP/voipjet-00708330", "1") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:4] Set("SIP/voipjet-00708330", "soundfile=DEBUG-20070314190006-1173913197.40.wav") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-00708330", "DEBUG-20070314190006-1173913197.40.wav") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:6] Set("SIP/voipjet-00708330", "CDR(accountcode)=477") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:7] Set("SIP/voipjet-00708330", "CDR(userfield)= 5083282553") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-00708330", "machinedetect") in new stack [Mar 14 19:00:06] -- Executing [s@liveonly:9] AMD("SIP/voipjet-00708330", "") in new stack [Mar 14 19:00:06] -- AMD: SIP/voipjet-00708330 6025161069 (null) (Fmt: 4) [Mar 14 19:00:06] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 19:00:07] -- AMD: Word detected. iWordsCount:1 [Mar 14 19:00:07] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 19:00:08] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 19:00:08] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-00708330", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 19:00:08] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-00708330", "0?liveonly-amd|s|1") in new stack [Mar 14 19:00:08] -- Executing [s@liveonly:12] Goto("SIP/voipjet-00708330", "liveonly-live|s|1") in new stack [Mar 14 19:00:08] -- Goto (liveonly-live,s,1) [Mar 14 19:00:08] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-00708330", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 19:00:08] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 19:00:16] == CDR updated on SIP/voipjet-00708330 [Mar 14 19:00:16] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-00708330", "Press 1 transfer") in new stack [Mar 14 19:00:16] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-00708330", "liveonly-transfer|s|1") in new stack [Mar 14 19:00:16] -- Goto (liveonly-transfer,s,1) [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-00708330", "soundfile=20070314190016-1173913197.40.ulaw") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-00708330", "20070314190016-1173913197.40.ulaw|b") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CALLERID(num)=3024573281") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-00708330", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-00708330", "CDR(userfield5)=2007-03-14 19:00:16") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-00708330", "0?20") in new stack [Mar 14 19:00:16] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-00708330", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 19:00:16] == Begin MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:00:16] RTP-stats [Mar 14 19:00:16] * Our Receiver: [Mar 14 19:00:16] SSRC: 0 [Mar 14 19:00:16] Received packets: 0 [Mar 14 19:00:16] Lost packets: 0 [Mar 14 19:00:16] Jitter: 0.0000 [Mar 14 19:00:16] Transit: 0.0000 [Mar 14 19:00:16] RR-count: 0 [Mar 14 19:00:16] * Our Sender: [Mar 14 19:00:16] SSRC: 9713593 [Mar 14 19:00:16] Sent packets: 0 [Mar 14 19:00:16] Lost packets: 0 [Mar 14 19:00:16] Jitter: 0 [Mar 14 19:00:16] SR-count: 0 [Mar 14 19:00:16] RTT: 0.000000 [Mar 14 19:00:16] Audio is at 167.206.178.7 port 18026 [Mar 14 19:00:16] Adding codec 0x4 (ulaw) to SDP [Mar 14 19:00:16] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 19:00:16] 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=z9hG4bK7486229f;rport From: "3024573281" ;tag=as3b31978b To: Contact: Call-ID: 68363db901da270f3b0cfbbe64b9a94a@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:00:16 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 18026 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 19:00:16] -- Called 13024573281@voipjet [Mar 14 19:00:16] -- Started music on hold, class 'trans3', on SIP/voipjet-00708330 [Mar 14 19:00:16] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK7486229f;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3b31978b To: Call-ID: 68363db901da270f3b0cfbbe64b9a94a@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 19:00:16] --- (10 headers 0 lines) --- [Mar 14 19:00:18] <--- 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=z9hG4bK7486229f;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3b31978b To: ;tag=as3a310c97 Call-ID: 68363db901da270f3b0cfbbe64b9a94a@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 17378 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 19:00:18] --- (11 headers 10 lines) --- [Mar 14 19:00:18] Found RTP audio format 0 [Mar 14 19:00:18] Found RTP audio format 101 [Mar 14 19:00:18] Peer audio RTP is at port 69.16.233.35:17378 [Mar 14 19:00:18] Found description format PCMU for ID 0 [Mar 14 19:00:18] Found description format telephone-event for ID 101 [Mar 14 19:00:18] Got unsupported a:fmtp in SDP offer [Mar 14 19:00:18] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:00:18] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:00:18] Peer audio RTP is at port 69.16.233.35:17378 [Mar 14 19:00:18] -- Call on SIP/voipjet-ae00b330 left from hold [Mar 14 19:00:18] -- Stopped music on hold on SIP/voipjet-00708330 [Mar 14 19:00:18] -- SIP/voipjet-ae00b330 is making progress passing it to SIP/voipjet-00708330 [Mar 14 19:00:21] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK7486229f;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3b31978b To: ;tag=as3a310c97 Call-ID: 68363db901da270f3b0cfbbe64b9a94a@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 17378 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 19:00:21] --- (11 headers 10 lines) --- [Mar 14 19:00:21] Found RTP audio format 0 [Mar 14 19:00:21] Found RTP audio format 101 [Mar 14 19:00:21] Peer audio RTP is at port 69.16.233.35:17378 [Mar 14 19:00:21] Found description format PCMU for ID 0 [Mar 14 19:00:21] Found description format telephone-event for ID 101 [Mar 14 19:00:21] Got unsupported a:fmtp in SDP offer [Mar 14 19:00:21] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 19:00:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 19:00:21] Peer audio RTP is at port 69.16.233.35:17378 [Mar 14 19:00:21] list_route: hop: [Mar 14 19:00:21] set_destination: Parsing for address/port to send to [Mar 14 19:00:21] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:00:21] 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=z9hG4bK13ce951a;rport From: "3024573281" ;tag=as3b31978b To: ;tag=as3a310c97 Contact: Call-ID: 68363db901da270f3b0cfbbe64b9a94a@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:00:21] -- Call on SIP/voipjet-ae00b330 left from hold [Mar 14 19:00:21] -- SIP/voipjet-ae00b330 answered SIP/voipjet-00708330 [Mar 14 19:00:32] 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=z9hG4bK06b1a239;rport From: "asterisk" ;tag=as0c940f59 To: Contact: Call-ID: 0809ef765185aeed7627c6730cd5deb9@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:00:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:00:32] <--- 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=z9hG4bK06b1a239;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as0c940f59 To: ;tag=as641a9e33 Call-ID: 0809ef765185aeed7627c6730cd5deb9@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 19:00:32] --- (10 headers 0 lines) --- [Mar 14 19:00:32] Really destroying SIP dialog '0809ef765185aeed7627c6730cd5deb9@167.206.178.7' Method: OPTIONS [Mar 14 19:00:32] 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=z9hG4bK677dc8f1;rport From: "asterisk" ;tag=as1fbbd61d To: Contact: Call-ID: 6fa25f3f1173e9c27d8f56bb1f8c8ea9@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:00:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:00:32] <--- 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=z9hG4bK677dc8f1;rport From: "asterisk" ;tag=as1fbbd61d To: Call-ID: 6fa25f3f1173e9c27d8f56bb1f8c8ea9@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 19:00:32] --- (7 headers 0 lines) --- [Mar 14 19:00:32] Really destroying SIP dialog '6fa25f3f1173e9c27d8f56bb1f8c8ea9@167.206.178.7' Method: OPTIONS [Mar 14 19:00:33] 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=z9hG4bK365cc823;rport From: "asterisk" ;tag=as42e35338 To: Contact: Call-ID: 51c3d7625afa05310db0400c6a03f4bd@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 23:00:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 19:00:33] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK365cc823;rport=5060 From: "asterisk" ;tag=as42e35338 To: ;tag=0-tdb3221212128 Call-ID: 51c3d7625afa05310db0400c6a03f4bd@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 19:00:33] --- (9 headers 0 lines) --- [Mar 14 19:00:33] Really destroying SIP dialog '51c3d7625afa05310db0400c6a03f4bd@167.206.178.7' Method: OPTIONS [Mar 14 19:00:34] <--- 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=z9hG4bK1638c91a;rport From: ;tag=as2747d21c To: "6025161069" ;tag=as2a076c81 Call-ID: 40c041d838afa4f41165ad9421b604cc@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 19:00:34] --- (9 headers 0 lines) --- [Mar 14 19:00:34] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 19:00:34] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK1638c91a;received=69.16.233.35;rport=5060 From: ;tag=as2747d21c To: "6025161069" ;tag=as2a076c81 Call-ID: 40c041d838afa4f41165ad9421b604cc@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 19:00:34] Scheduling destruction of SIP dialog '68363db901da270f3b0cfbbe64b9a94a@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 19:00:34] set_destination: Parsing for address/port to send to [Mar 14 19:00:34] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 19:00:34] 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=z9hG4bK65977840;rport From: "3024573281" ;tag=as3b31978b To: ;tag=as3a310c97 Call-ID: 68363db901da270f3b0cfbbe64b9a94a@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 19:00:34] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-00708330' [Mar 14 19:00:34] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-00708330", "CDR(userfield)=press1transfer 20070314190016-1173913197.40.ulaw") in new stack [Mar 14 19:00:34] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-00708330", "CDR(userfield2)=5083282553") in new stack [Mar 14 19:00:34] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-00708330", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 19:00:34] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-00708330", "CDR(accountcode)=477") in new stack [Mar 14 19:00:34] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-00708330", "5083282553") in new stack [Mar 14 19:00:34] == End MixMonitor Recording SIP/voipjet-00708330 [Mar 14 19:00:34] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK65977840;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as3b31978b To: ;tag=as3a310c97 Call-ID: 68363db901da270f3b0cfbbe64b9a94a@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 19:00:34] --- (11 headers 0 lines) --- [Mar 14 19:00:34] Really destroying SIP dialog '68363db901da270f3b0cfbbe64b9a94a@167.206.178.7' Method: INVITE [Mar 14 19:00:34] RTP-stats [Mar 14 19:00:34] * Our Receiver: [Mar 14 19:00:34] SSRC: 255342345 [Mar 14 19:00:34] Received packets: 784 [Mar 14 19:00:34] Lost packets: 0 [Mar 14 19:00:34] Jitter: 0.0003 [Mar 14 19:00:34] Transit: -0.0003 [Mar 14 19:00:34] RR-count: 0 [Mar 14 19:00:34] * Our Sender: [Mar 14 19:00:34] SSRC: 201788722 [Mar 14 19:00:34] Sent packets: 417 [Mar 14 19:00:34] Lost packets: 0 [Mar 14 19:00:34] Jitter: 0 [Mar 14 19:00:34] SR-count: 3 [Mar 14 19:00:34] RTT: 0.000000 [Mar 14 19:00:34] Really destroying SIP dialog '40c041d838afa4f41165ad9421b604cc@167.206.178.7' Method: BYE [Mar 14 19:00:34] RTP-stats [Mar 14 19:00:34] * Our Receiver: [Mar 14 19:00:34] SSRC: 68806544 [Mar 14 19:00:34] Received packets: 1305 [Mar 14 19:00:34] Lost packets: 6 [Mar 14 19:00:34] Jitter: 0.0001 [Mar 14 19:00:34] Transit: -0.0001 [Mar 14 19:00:34] RR-count: 1 [Mar 14 19:00:34] * Our Sender: [Mar 14 19:00:34] SSRC: 1987734217 [Mar 14 19:00:34] Sent packets: 1116 [Mar 14 19:00:34] Lost packets: 0 [Mar 14 19:00:34] Jitter: 0 [Mar 14 19:00:34] SR-count: 5 [Mar 14 19:00:34] RTT: 0.000000