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 18:41:37] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:41:37] Found [Mar 14 18:41:40] 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=z9hG4bK0c810480;rport From: "asterisk" ;tag=as041ff75f To: Contact: Call-ID: 395e4578715475176581717d07147851@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:41:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:41:40] <--- 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=z9hG4bK0c810480;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as041ff75f To: ;tag=as02634858 Call-ID: 395e4578715475176581717d07147851@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 18:41:40] --- (10 headers 0 lines) --- [Mar 14 18:41:40] Really destroying SIP dialog '395e4578715475176581717d07147851@167.206.178.7' Method: OPTIONS [Mar 14 18:41:41] 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=z9hG4bK7ca696c4;rport From: "asterisk" ;tag=as2d682d9c To: Contact: Call-ID: 63b01d1977d2fcc5437b6ebb30b8d5a2@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:41:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:41:41] <--- 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=z9hG4bK7ca696c4;rport From: "asterisk" ;tag=as2d682d9c To: Call-ID: 63b01d1977d2fcc5437b6ebb30b8d5a2@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 18:41:41] --- (7 headers 0 lines) --- [Mar 14 18:41:41] Really destroying SIP dialog '63b01d1977d2fcc5437b6ebb30b8d5a2@167.206.178.7' Method: OPTIONS [Mar 14 18:41:42] 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=z9hG4bK51a7938f;rport From: "asterisk" ;tag=as2ed75c89 To: Contact: Call-ID: 622d580c1e27afbd42f2bffc0d6b2036@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:41:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:41:42] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK51a7938f;rport=5060 From: "asterisk" ;tag=as2ed75c89 To: ;tag=0-tdb3221212128 Call-ID: 622d580c1e27afbd42f2bffc0d6b2036@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 18:41:42] --- (9 headers 0 lines) --- [Mar 14 18:41:42] Really destroying SIP dialog '622d580c1e27afbd42f2bffc0d6b2036@167.206.178.7' Method: OPTIONS [Mar 14 18:41:56] RTP-stats [Mar 14 18:41:56] * Our Receiver: [Mar 14 18:41:56] SSRC: 0 [Mar 14 18:41:56] Received packets: 0 [Mar 14 18:41:56] Lost packets: 0 [Mar 14 18:41:56] Jitter: 0.0000 [Mar 14 18:41:56] Transit: 0.0000 [Mar 14 18:41:56] RR-count: 0 [Mar 14 18:41:56] * Our Sender: [Mar 14 18:41:56] SSRC: 1924896667 [Mar 14 18:41:56] Sent packets: 0 [Mar 14 18:41:56] Lost packets: 0 [Mar 14 18:41:56] Jitter: 0 [Mar 14 18:41:56] SR-count: 0 [Mar 14 18:41:56] RTT: 0.000000 [Mar 14 18:41:56] Audio is at 167.206.178.7 port 10678 [Mar 14 18:41:56] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:41:56] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:41:56] 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=z9hG4bK41ac2003;rport From: "6025161069" ;tag=as48d5339f To: Contact: Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:41:56 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 10678 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 18:41:56] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK41ac2003;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as48d5339f To: Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@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 18:41:56] --- (10 headers 0 lines) --- [Mar 14 18:42:01] <--- 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=z9hG4bK41ac2003;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as48d5339f To: ;tag=as0b672f04 Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@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 18860 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 18:42:01] --- (11 headers 10 lines) --- [Mar 14 18:42:01] Found RTP audio format 0 [Mar 14 18:42:01] Found RTP audio format 101 [Mar 14 18:42:01] Peer audio RTP is at port 69.16.233.35:18860 [Mar 14 18:42:01] Found description format PCMU for ID 0 [Mar 14 18:42:01] Found description format telephone-event for ID 101 [Mar 14 18:42:01] Got unsupported a:fmtp in SDP offer [Mar 14 18:42:01] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:42:01] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:42:01] Peer audio RTP is at port 69.16.233.35:18860 [Mar 14 18:42:04] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK41ac2003;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as48d5339f To: ;tag=as0b672f04 Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@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 18860 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 18:42:04] --- (11 headers 10 lines) --- [Mar 14 18:42:04] Found RTP audio format 0 [Mar 14 18:42:04] Found RTP audio format 101 [Mar 14 18:42:04] Peer audio RTP is at port 69.16.233.35:18860 [Mar 14 18:42:04] Found description format PCMU for ID 0 [Mar 14 18:42:04] Found description format telephone-event for ID 101 [Mar 14 18:42:04] Got unsupported a:fmtp in SDP offer [Mar 14 18:42:04] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:42:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:42:04] Peer audio RTP is at port 69.16.233.35:18860 [Mar 14 18:42:04] list_route: hop: [Mar 14 18:42:04] set_destination: Parsing for address/port to send to [Mar 14 18:42:04] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:42:04] 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=z9hG4bK7c98c2bf;rport From: "6025161069" ;tag=as48d5339f To: ;tag=as0b672f04 Contact: Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:42:04] > Channel SIP/voipjet-0070afe0 was answered. [Mar 14 18:42:04] -- Executing [s@liveonly:1] Answer("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:42:04] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-0070afe0", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:474 CALLERID: ") in new stack [Mar 14 18:42:04] -- Executing [s@liveonly:3] Wait("SIP/voipjet-0070afe0", "1") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:4] Set("SIP/voipjet-0070afe0", "soundfile=DEBUG-20070314184206-1173912116.32.wav") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-0070afe0", "DEBUG-20070314184206-1173912116.32.wav") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:6] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=474") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:7] Set("SIP/voipjet-0070afe0", "CDR(userfield)= 5083282553") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-0070afe0", "machinedetect") in new stack [Mar 14 18:42:06] -- Executing [s@liveonly:9] AMD("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:42:06] -- AMD: SIP/voipjet-0070afe0 6025161069 (null) (Fmt: 4) [Mar 14 18:42:06] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 18:42:07] -- AMD: Word detected. iWordsCount:1 [Mar 14 18:42:08] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 18:42:09] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 18:42:09] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-0070afe0", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 18:42:09] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-0070afe0", "0?liveonly-amd|s|1") in new stack [Mar 14 18:42:09] -- Executing [s@liveonly:12] Goto("SIP/voipjet-0070afe0", "liveonly-live|s|1") in new stack [Mar 14 18:42:09] -- Goto (liveonly-live,s,1) [Mar 14 18:42:09] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-0070afe0", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 18:42:09] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 18:42:12] == CDR updated on SIP/voipjet-0070afe0 [Mar 14 18:42:12] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-0070afe0", "Press 1 transfer") in new stack [Mar 14 18:42:12] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-0070afe0", "liveonly-transfer|s|1") in new stack [Mar 14 18:42:12] -- Goto (liveonly-transfer,s,1) [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "soundfile=20070314184212-1173912116.32.ulaw") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-0070afe0", "20070314184212-1173912116.32.ulaw|b") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CALLERID(num)=3024573281") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-0070afe0", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-0070afe0", "CDR(userfield5)=2007-03-14 18:42:12") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-0070afe0", "0?20") in new stack [Mar 14 18:42:12] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-0070afe0", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 18:42:12] RTP-stats [Mar 14 18:42:12] * Our Receiver: [Mar 14 18:42:12] SSRC: 0 [Mar 14 18:42:12] Received packets: 0 [Mar 14 18:42:12] Lost packets: 0 [Mar 14 18:42:12] Jitter: 0.0000 [Mar 14 18:42:12] Transit: 0.0000 [Mar 14 18:42:12] == Begin MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:42:12] RR-count: 0 [Mar 14 18:42:12] * Our Sender: [Mar 14 18:42:12] SSRC: 1217367486 [Mar 14 18:42:12] Sent packets: 0 [Mar 14 18:42:12] Lost packets: 0 [Mar 14 18:42:12] Jitter: 0 [Mar 14 18:42:12] SR-count: 0 [Mar 14 18:42:12] RTT: 0.000000 [Mar 14 18:42:12] Audio is at 167.206.178.7 port 13678 [Mar 14 18:42:12] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:42:12] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:42:12] 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=z9hG4bK19c36780;rport From: "3024573281" ;tag=as4f3b299b To: Contact: Call-ID: 2301839f2a15fbf037f7368c162b7485@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:42:12 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 13678 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 18:42:12] -- Called 13024573281@voipjet [Mar 14 18:42:12] -- Started music on hold, class 'trans3', on SIP/voipjet-0070afe0 [Mar 14 18:42:12] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK19c36780;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4f3b299b To: Call-ID: 2301839f2a15fbf037f7368c162b7485@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 18:42:12] --- (10 headers 0 lines) --- [Mar 14 18:42:14] <--- 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=z9hG4bK19c36780;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4f3b299b To: ;tag=as2d02f6d2 Call-ID: 2301839f2a15fbf037f7368c162b7485@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 14578 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 18:42:14] --- (11 headers 10 lines) --- [Mar 14 18:42:14] Found RTP audio format 0 [Mar 14 18:42:14] Found RTP audio format 101 [Mar 14 18:42:14] Peer audio RTP is at port 69.16.233.35:14578 [Mar 14 18:42:14] Found description format PCMU for ID 0 [Mar 14 18:42:14] Found description format telephone-event for ID 101 [Mar 14 18:42:14] Got unsupported a:fmtp in SDP offer [Mar 14 18:42:14] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:42:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:42:14] Peer audio RTP is at port 69.16.233.35:14578 [Mar 14 18:42:14] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:42:14] -- Stopped music on hold on SIP/voipjet-0070afe0 [Mar 14 18:42:14] -- SIP/voipjet-006c0ff0 is making progress passing it to SIP/voipjet-0070afe0 [Mar 14 18:42:17] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK19c36780;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4f3b299b To: ;tag=as2d02f6d2 Call-ID: 2301839f2a15fbf037f7368c162b7485@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 14578 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 18:42:17] --- (11 headers 10 lines) --- [Mar 14 18:42:17] Found RTP audio format 0 [Mar 14 18:42:17] Found RTP audio format 101 [Mar 14 18:42:17] Peer audio RTP is at port 69.16.233.35:14578 [Mar 14 18:42:17] Found description format PCMU for ID 0 [Mar 14 18:42:17] Found description format telephone-event for ID 101 [Mar 14 18:42:17] Got unsupported a:fmtp in SDP offer [Mar 14 18:42:17] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:42:17] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:42:17] Peer audio RTP is at port 69.16.233.35:14578 [Mar 14 18:42:17] list_route: hop: [Mar 14 18:42:17] set_destination: Parsing for address/port to send to [Mar 14 18:42:17] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:42:17] 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=z9hG4bK358beeb6;rport From: "3024573281" ;tag=as4f3b299b To: ;tag=as2d02f6d2 Contact: Call-ID: 2301839f2a15fbf037f7368c162b7485@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:42:17] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:42:17] -- SIP/voipjet-006c0ff0 answered SIP/voipjet-0070afe0 [Mar 14 18:42:26] <--- 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=z9hG4bK364f3fde;rport From: ;tag=as0b672f04 To: "6025161069" ;tag=as48d5339f Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 18:42:26] --- (9 headers 0 lines) --- [Mar 14 18:42:26] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 18:42:26] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK364f3fde;received=69.16.233.35;rport=5060 From: ;tag=as0b672f04 To: "6025161069" ;tag=as48d5339f Call-ID: 5c2a74f7088dc857313cb77e2fd736b5@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 18:42:26] Scheduling destruction of SIP dialog '2301839f2a15fbf037f7368c162b7485@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 18:42:26] set_destination: Parsing for address/port to send to [Mar 14 18:42:26] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:42:26] 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=z9hG4bK243b9d49;rport From: "3024573281" ;tag=as4f3b299b To: ;tag=as2d02f6d2 Call-ID: 2301839f2a15fbf037f7368c162b7485@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:42:26] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-0070afe0' [Mar 14 18:42:26] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "CDR(userfield)=press1transfer 20070314184212-1173912116.32.ulaw") in new stack [Mar 14 18:42:26] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-0070afe0", "CDR(userfield2)=5083282553") in new stack [Mar 14 18:42:26] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 18:42:26] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=474") in new stack [Mar 14 18:42:26] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-0070afe0", "5083282553") in new stack [Mar 14 18:42:26] == End MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:42:26] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK243b9d49;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as4f3b299b To: ;tag=as2d02f6d2 Call-ID: 2301839f2a15fbf037f7368c162b7485@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 18:42:26] --- (11 headers 0 lines) --- [Mar 14 18:42:26] Really destroying SIP dialog '2301839f2a15fbf037f7368c162b7485@167.206.178.7' Method: INVITE [Mar 14 18:42:26] RTP-stats [Mar 14 18:42:26] * Our Receiver: [Mar 14 18:42:26] SSRC: 1615071423 [Mar 14 18:42:26] Received packets: 597 [Mar 14 18:42:26] Lost packets: 2 [Mar 14 18:42:26] Jitter: 0.0006 [Mar 14 18:42:26] Transit: -0.0002 [Mar 14 18:42:26] RR-count: 0 [Mar 14 18:42:26] * Our Sender: [Mar 14 18:42:26] SSRC: 1982627470 [Mar 14 18:42:26] Sent packets: 422 [Mar 14 18:42:26] Lost packets: 0 [Mar 14 18:42:26] Jitter: 0 [Mar 14 18:42:26] SR-count: 2 [Mar 14 18:42:26] RTT: 0.000000 [Mar 14 18:42:26] Really destroying SIP dialog '5c2a74f7088dc857313cb77e2fd736b5@167.206.178.7' Method: BYE [Mar 14 18:42:26] RTP-stats [Mar 14 18:42:26] * Our Receiver: [Mar 14 18:42:26] SSRC: 235239702 [Mar 14 18:42:26] Received packets: 1199 [Mar 14 18:42:26] Lost packets: 7 [Mar 14 18:42:26] Jitter: 0.0001 [Mar 14 18:42:26] Transit: -0.0002 [Mar 14 18:42:26] RR-count: 1 [Mar 14 18:42:26] * Our Sender: [Mar 14 18:42:26] SSRC: 1203092100 [Mar 14 18:42:26] Sent packets: 696 [Mar 14 18:42:26] Lost packets: 0 [Mar 14 18:42:26] Jitter: 0 [Mar 14 18:42:26] SR-count: 3 [Mar 14 18:42:26] RTT: 0.000000