ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> [Mar 14 18:39:37] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:39:37] Found [Mar 14 18:39:40] 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=z9hG4bK0501d422;rport From: "asterisk" ;tag=as3c36ccc6 To: Contact: Call-ID: 38938c032d291e363d91a8802b53a3dc@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:39:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:39:40] <--- 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=z9hG4bK0501d422;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as3c36ccc6 To: ;tag=as490c0810 Call-ID: 38938c032d291e363d91a8802b53a3dc@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 18:39:40] --- (10 headers 0 lines) --- [Mar 14 18:39:40] Really destroying SIP dialog '38938c032d291e363d91a8802b53a3dc@167.206.178.7' Method: OPTIONS [Mar 14 18:39:41] 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=z9hG4bK73487ef5;rport From: "asterisk" ;tag=as37859249 To: Contact: Call-ID: 4f1c014f23a66a945ef1187919c63be0@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:39:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:39:41] <--- 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=z9hG4bK73487ef5;rport From: "asterisk" ;tag=as37859249 To: Call-ID: 4f1c014f23a66a945ef1187919c63be0@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 18:39:41] --- (7 headers 0 lines) --- [Mar 14 18:39:41] Really destroying SIP dialog '4f1c014f23a66a945ef1187919c63be0@167.206.178.7' Method: OPTIONS [Mar 14 18:39:42] 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=z9hG4bK54e16b02;rport From: "asterisk" ;tag=as68d0b88d To: Contact: Call-ID: 272d103a42a450b82ce961180ca49410@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:39:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:39:42] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK54e16b02;rport=5060 From: "asterisk" ;tag=as68d0b88d To: ;tag=0-tdb3221212128 Call-ID: 272d103a42a450b82ce961180ca49410@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 18:39:42] --- (9 headers 0 lines) --- [Mar 14 18:39:42] Really destroying SIP dialog '272d103a42a450b82ce961180ca49410@167.206.178.7' Method: OPTIONS [Mar 14 18:39:56] RTP-stats [Mar 14 18:39:56] * Our Receiver: [Mar 14 18:39:56] SSRC: 0 [Mar 14 18:39:56] Received packets: 0 [Mar 14 18:39:56] Lost packets: 0 [Mar 14 18:39:56] Jitter: 0.0000 [Mar 14 18:39:56] Transit: 0.0000 [Mar 14 18:39:56] RR-count: 0 [Mar 14 18:39:56] * Our Sender: [Mar 14 18:39:56] SSRC: 599538022 [Mar 14 18:39:56] Sent packets: 0 [Mar 14 18:39:56] Lost packets: 0 [Mar 14 18:39:56] Jitter: 0 [Mar 14 18:39:56] SR-count: 0 [Mar 14 18:39:56] RTT: 0.000000 [Mar 14 18:39:56] Audio is at 167.206.178.7 port 14516 [Mar 14 18:39:56] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:39:56] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:39:56] 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=z9hG4bK73716c56;rport From: "6025161069" ;tag=as71571cbf To: Contact: Call-ID: 52fa222b2895efbc6094232e445081a4@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:39:56 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 14516 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 18:39:56] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK73716c56;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as71571cbf To: Call-ID: 52fa222b2895efbc6094232e445081a4@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 18:39:56] --- (10 headers 0 lines) --- [Mar 14 18:40:02] <--- 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=z9hG4bK73716c56;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as71571cbf To: ;tag=as50c2b297 Call-ID: 52fa222b2895efbc6094232e445081a4@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 15114 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 18:40:02] --- (11 headers 10 lines) --- [Mar 14 18:40:02] Found RTP audio format 0 [Mar 14 18:40:02] Found RTP audio format 101 [Mar 14 18:40:02] Peer audio RTP is at port 69.16.233.35:15114 [Mar 14 18:40:02] Found description format PCMU for ID 0 [Mar 14 18:40:02] Found description format telephone-event for ID 101 [Mar 14 18:40:02] Got unsupported a:fmtp in SDP offer [Mar 14 18:40:02] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:40:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:40:02] Peer audio RTP is at port 69.16.233.35:15114 [Mar 14 18:40:15] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK73716c56;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as71571cbf To: ;tag=as50c2b297 Call-ID: 52fa222b2895efbc6094232e445081a4@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 15114 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 18:40:15] --- (11 headers 10 lines) --- [Mar 14 18:40:15] Found RTP audio format 0 [Mar 14 18:40:15] Found RTP audio format 101 [Mar 14 18:40:15] Peer audio RTP is at port 69.16.233.35:15114 [Mar 14 18:40:15] Found description format PCMU for ID 0 [Mar 14 18:40:15] Found description format telephone-event for ID 101 [Mar 14 18:40:15] Got unsupported a:fmtp in SDP offer [Mar 14 18:40:15] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:40:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:40:15] Peer audio RTP is at port 69.16.233.35:15114 [Mar 14 18:40:15] list_route: hop: [Mar 14 18:40:15] set_destination: Parsing for address/port to send to [Mar 14 18:40:15] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:40:15] 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=z9hG4bK02956fcc;rport From: "6025161069" ;tag=as71571cbf To: ;tag=as50c2b297 Contact: Call-ID: 52fa222b2895efbc6094232e445081a4@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:40:15] > Channel SIP/voipjet-0070afe0 was answered. [Mar 14 18:40:15] -- Executing [s@liveonly:1] Answer("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:40:15] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-0070afe0", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:473 CALLERID: ") in new stack [Mar 14 18:40:15] -- Executing [s@liveonly:3] Wait("SIP/voipjet-0070afe0", "1") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:4] Set("SIP/voipjet-0070afe0", "soundfile=DEBUG-20070314184016-1173911996.30.wav") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-0070afe0", "DEBUG-20070314184016-1173911996.30.wav") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:6] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=473") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:7] Set("SIP/voipjet-0070afe0", "CDR(userfield)= 5083282553") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-0070afe0", "machinedetect") in new stack [Mar 14 18:40:16] -- Executing [s@liveonly:9] AMD("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:40:16] -- AMD: SIP/voipjet-0070afe0 6025161069 (null) (Fmt: 4) [Mar 14 18:40:16] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 18:40:19] -- AMD: Word detected. iWordsCount:1 [Mar 14 18:40:19] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 18:40:20] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 18:40:20] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-0070afe0", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 18:40:20] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-0070afe0", "0?liveonly-amd|s|1") in new stack [Mar 14 18:40:20] -- Executing [s@liveonly:12] Goto("SIP/voipjet-0070afe0", "liveonly-live|s|1") in new stack [Mar 14 18:40:20] -- Goto (liveonly-live,s,1) [Mar 14 18:40:20] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-0070afe0", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 18:40:20] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 18:40:24] == CDR updated on SIP/voipjet-0070afe0 [Mar 14 18:40:24] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-0070afe0", "Press 1 transfer") in new stack [Mar 14 18:40:24] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-0070afe0", "liveonly-transfer|s|1") in new stack [Mar 14 18:40:24] -- Goto (liveonly-transfer,s,1) [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "soundfile=20070314184024-1173911996.30.ulaw") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-0070afe0", "20070314184024-1173911996.30.ulaw|b") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CALLERID(num)=3024573281") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-0070afe0", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-0070afe0", "CDR(userfield5)=2007-03-14 18:40:24") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-0070afe0", "0?20") in new stack [Mar 14 18:40:24] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-0070afe0", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 18:40:24] RTP-stats [Mar 14 18:40:24] * Our Receiver: [Mar 14 18:40:24] == Begin MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:40:24] SSRC: 0 [Mar 14 18:40:24] Received packets: 0 [Mar 14 18:40:24] Lost packets: 0 [Mar 14 18:40:24] Jitter: 0.0000 [Mar 14 18:40:24] Transit: 0.0000 [Mar 14 18:40:24] RR-count: 0 [Mar 14 18:40:24] * Our Sender: [Mar 14 18:40:24] SSRC: 1949105254 [Mar 14 18:40:24] Sent packets: 0 [Mar 14 18:40:24] Lost packets: 0 [Mar 14 18:40:24] Jitter: 0 [Mar 14 18:40:24] SR-count: 0 [Mar 14 18:40:24] RTT: 0.000000 [Mar 14 18:40:24] Audio is at 167.206.178.7 port 10926 [Mar 14 18:40:24] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:40:24] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:40:24] 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=z9hG4bK35ad09a3;rport From: "3024573281" ;tag=as1c5eb35f To: Contact: Call-ID: 7282425d1dd5e6396937b1611422dfac@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:40:24 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 10926 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 18:40:24] -- Called 13024573281@voipjet [Mar 14 18:40:24] -- Started music on hold, class 'trans3', on SIP/voipjet-0070afe0 [Mar 14 18:40:24] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK35ad09a3;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as1c5eb35f To: Call-ID: 7282425d1dd5e6396937b1611422dfac@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 18:40:24] --- (10 headers 0 lines) --- [Mar 14 18:40:26] <--- 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=z9hG4bK35ad09a3;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as1c5eb35f To: ;tag=as5727904f Call-ID: 7282425d1dd5e6396937b1611422dfac@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 13470 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 18:40:26] --- (11 headers 10 lines) --- [Mar 14 18:40:26] Found RTP audio format 0 [Mar 14 18:40:26] Found RTP audio format 101 [Mar 14 18:40:26] Peer audio RTP is at port 69.16.233.35:13470 [Mar 14 18:40:26] Found description format PCMU for ID 0 [Mar 14 18:40:26] Found description format telephone-event for ID 101 [Mar 14 18:40:26] Got unsupported a:fmtp in SDP offer [Mar 14 18:40:26] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:40:26] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:40:26] Peer audio RTP is at port 69.16.233.35:13470 [Mar 14 18:40:26] -- Call on SIP/voipjet-ae01a030 left from hold [Mar 14 18:40:26] -- Stopped music on hold on SIP/voipjet-0070afe0 [Mar 14 18:40:26] -- SIP/voipjet-ae01a030 is making progress passing it to SIP/voipjet-0070afe0 [Mar 14 18:40:29] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK35ad09a3;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as1c5eb35f To: ;tag=as5727904f Call-ID: 7282425d1dd5e6396937b1611422dfac@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 13470 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 18:40:29] --- (11 headers 10 lines) --- [Mar 14 18:40:29] Found RTP audio format 0 [Mar 14 18:40:29] Found RTP audio format 101 [Mar 14 18:40:29] Peer audio RTP is at port 69.16.233.35:13470 [Mar 14 18:40:29] Found description format PCMU for ID 0 [Mar 14 18:40:29] Found description format telephone-event for ID 101 [Mar 14 18:40:29] Got unsupported a:fmtp in SDP offer [Mar 14 18:40:29] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:40:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:40:29] Peer audio RTP is at port 69.16.233.35:13470 [Mar 14 18:40:29] list_route: hop: [Mar 14 18:40:29] set_destination: Parsing for address/port to send to [Mar 14 18:40:29] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:40:29] 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=z9hG4bK3b218b19;rport From: "3024573281" ;tag=as1c5eb35f To: ;tag=as5727904f Contact: Call-ID: 7282425d1dd5e6396937b1611422dfac@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:40:29] -- Call on SIP/voipjet-ae01a030 left from hold [Mar 14 18:40:29] -- SIP/voipjet-ae01a030 answered SIP/voipjet-0070afe0 [Mar 14 18:40:39] <--- 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=z9hG4bK4a4030fb;rport From: ;tag=as50c2b297 To: "6025161069" ;tag=as71571cbf Call-ID: 52fa222b2895efbc6094232e445081a4@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 18:40:39] --- (9 headers 0 lines) --- [Mar 14 18:40:39] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 18:40:39] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK4a4030fb;received=69.16.233.35;rport=5060 From: ;tag=as50c2b297 To: "6025161069" ;tag=as71571cbf Call-ID: 52fa222b2895efbc6094232e445081a4@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 18:40:39] Scheduling destruction of SIP dialog '7282425d1dd5e6396937b1611422dfac@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 18:40:39] set_destination: Parsing for address/port to send to [Mar 14 18:40:39] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:40:39] 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=z9hG4bK00dbdca8;rport From: "3024573281" ;tag=as1c5eb35f To: ;tag=as5727904f Call-ID: 7282425d1dd5e6396937b1611422dfac@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:40:39] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-0070afe0' [Mar 14 18:40:39] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "CDR(userfield)=press1transfer 20070314184024-1173911996.30.ulaw") in new stack [Mar 14 18:40:39] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-0070afe0", "CDR(userfield2)=5083282553") in new stack [Mar 14 18:40:39] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 18:40:39] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=473") in new stack [Mar 14 18:40:39] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-0070afe0", "5083282553") in new stack [Mar 14 18:40:39] == End MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:40:39] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK00dbdca8;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as1c5eb35f To: ;tag=as5727904f Call-ID: 7282425d1dd5e6396937b1611422dfac@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 18:40:39] --- (11 headers 0 lines) --- [Mar 14 18:40:39] Really destroying SIP dialog '7282425d1dd5e6396937b1611422dfac@167.206.178.7' Method: INVITE [Mar 14 18:40:39] RTP-stats [Mar 14 18:40:39] * Our Receiver: [Mar 14 18:40:39] SSRC: 883508208 [Mar 14 18:40:39] Received packets: 644 [Mar 14 18:40:39] Lost packets: 1 [Mar 14 18:40:39] Jitter: 0.0001 [Mar 14 18:40:39] Transit: 0.0001 [Mar 14 18:40:39] RR-count: 0 [Mar 14 18:40:39] * Our Sender: [Mar 14 18:40:39] SSRC: 1063259695 [Mar 14 18:40:39] Sent packets: 479 [Mar 14 18:40:39] Lost packets: 0 [Mar 14 18:40:39] Jitter: 0 [Mar 14 18:40:39] SR-count: 2 [Mar 14 18:40:39] RTT: 0.000000 [Mar 14 18:40:39] Really destroying SIP dialog '52fa222b2895efbc6094232e445081a4@167.206.178.7' Method: BYE [Mar 14 18:40:39] RTP-stats [Mar 14 18:40:39] * Our Receiver: [Mar 14 18:40:39] SSRC: 721359833 [Mar 14 18:40:39] Received packets: 1811 [Mar 14 18:40:39] Lost packets: 0 [Mar 14 18:40:39] Jitter: 0.0002 [Mar 14 18:40:39] Transit: 0.0003 [Mar 14 18:40:39] RR-count: 3 [Mar 14 18:40:39] * Our Sender: [Mar 14 18:40:39] SSRC: 1583200375 [Mar 14 18:40:39] Sent packets: 780 [Mar 14 18:40:39] Lost packets: 0 [Mar 14 18:40:39] Jitter: 0 [Mar 14 18:40:39] SR-count: 4 [Mar 14 18:40:39] RTT: 0.000000