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 18:49:07] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:49:07] Found ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> ASTEISK05*CLI> [Mar 14 18:49:18] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:49:18] Found [Mar 14 18:49:23] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:49:23] Found [Mar 14 18:49:28] == Parsing '/etc/asterisk/manager.conf': [Mar 14 18:49:28] Found [Mar 14 18:49:38] RTP-stats [Mar 14 18:49:38] * Our Receiver: [Mar 14 18:49:38] SSRC: 0 [Mar 14 18:49:38] Received packets: 0 [Mar 14 18:49:38] Lost packets: 0 [Mar 14 18:49:38] Jitter: 0.0000 [Mar 14 18:49:38] Transit: 0.0000 [Mar 14 18:49:38] RR-count: 0 [Mar 14 18:49:38] * Our Sender: [Mar 14 18:49:38] SSRC: 1271240296 [Mar 14 18:49:38] Sent packets: 0 [Mar 14 18:49:38] Lost packets: 0 [Mar 14 18:49:38] Jitter: 0 [Mar 14 18:49:38] SR-count: 0 [Mar 14 18:49:38] RTT: 0.000000 [Mar 14 18:49:38] Audio is at 167.206.178.7 port 18180 [Mar 14 18:49:38] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:49:38] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:49:38] 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=z9hG4bK060960c8;rport From: "6025161069" ;tag=as49e679d2 To: Contact: Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:49:38 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 18180 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 18:49:38] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK060960c8;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as49e679d2 To: Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@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 18:49:38] --- (10 headers 0 lines) --- [Mar 14 18:49:40] 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=z9hG4bK181e031e;rport From: "asterisk" ;tag=as78b3080d To: Contact: Call-ID: 1583e06765a5adf23b16abf21d6ab7be@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:49:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:49:40] <--- 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=z9hG4bK181e031e;received=167.206.178.7;rport=5060 From: "asterisk" ;tag=as78b3080d To: ;tag=as424059f9 Call-ID: 1583e06765a5adf23b16abf21d6ab7be@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 18:49:40] --- (10 headers 0 lines) --- [Mar 14 18:49:40] Really destroying SIP dialog '1583e06765a5adf23b16abf21d6ab7be@167.206.178.7' Method: OPTIONS [Mar 14 18:49:41] 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=z9hG4bK48d0206e;rport From: "asterisk" ;tag=as6d0d9439 To: Contact: Call-ID: 1797b28042915807724385dd5918307c@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:49:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:49:41] <--- 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=z9hG4bK48d0206e;rport From: "asterisk" ;tag=as6d0d9439 To: Call-ID: 1797b28042915807724385dd5918307c@167.206.178.7 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [Mar 14 18:49:41] --- (7 headers 0 lines) --- [Mar 14 18:49:41] Really destroying SIP dialog '1797b28042915807724385dd5918307c@167.206.178.7' Method: OPTIONS [Mar 14 18:49:42] 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=z9hG4bK3dc4cdb7;rport From: "asterisk" ;tag=as7ffb07d8 To: Contact: Call-ID: 63d498aa2169af901dc3ab593b00dc30@167.206.178.7 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:49:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 14 18:49:43] <--- SIP read from 66.63.165.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK3dc4cdb7;rport=5060 From: "asterisk" ;tag=as7ffb07d8 To: ;tag=0-tdb3221212128 Call-ID: 63d498aa2169af901dc3ab593b00dc30@167.206.178.7 CSeq: 102 OPTIONS Server: Sansay VSX 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Length: 0 <-------------> [Mar 14 18:49:43] --- (9 headers 0 lines) --- [Mar 14 18:49:43] Really destroying SIP dialog '63d498aa2169af901dc3ab593b00dc30@167.206.178.7' Method: OPTIONS [Mar 14 18:49:43] <--- 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=z9hG4bK060960c8;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as49e679d2 To: ;tag=as5e37f444 Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@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 18668 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 18:49:43] --- (11 headers 10 lines) --- [Mar 14 18:49:43] Found RTP audio format 0 [Mar 14 18:49:43] Found RTP audio format 101 [Mar 14 18:49:43] Peer audio RTP is at port 69.16.233.35:18668 [Mar 14 18:49:43] Found description format PCMU for ID 0 [Mar 14 18:49:43] Found description format telephone-event for ID 101 [Mar 14 18:49:43] Got unsupported a:fmtp in SDP offer [Mar 14 18:49:43] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:49:43] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:49:43] Peer audio RTP is at port 69.16.233.35:18668 [Mar 14 18:49:49] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK060960c8;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as49e679d2 To: ;tag=as5e37f444 Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@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 18668 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 18:49:49] --- (11 headers 10 lines) --- [Mar 14 18:49:49] Found RTP audio format 0 [Mar 14 18:49:49] Found RTP audio format 101 [Mar 14 18:49:49] Peer audio RTP is at port 69.16.233.35:18668 [Mar 14 18:49:49] Found description format PCMU for ID 0 [Mar 14 18:49:49] Found description format telephone-event for ID 101 [Mar 14 18:49:49] Got unsupported a:fmtp in SDP offer [Mar 14 18:49:49] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:49:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:49:49] Peer audio RTP is at port 69.16.233.35:18668 [Mar 14 18:49:49] list_route: hop: [Mar 14 18:49:49] set_destination: Parsing for address/port to send to [Mar 14 18:49:49] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:49:49] 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=z9hG4bK0185084e;rport From: "6025161069" ;tag=as49e679d2 To: ;tag=as5e37f444 Contact: Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:49:49] > Channel SIP/voipjet-0070afe0 was answered. [Mar 14 18:49:49] -- Executing [s@liveonly:1] Answer("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:49:49] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-0070afe0", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:477 CALLERID: ") in new stack [Mar 14 18:49:49] -- Executing [s@liveonly:3] Wait("SIP/voipjet-0070afe0", "1") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:4] Set("SIP/voipjet-0070afe0", "soundfile=DEBUG-20070314184950-1173912578.38.wav") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-0070afe0", "DEBUG-20070314184950-1173912578.38.wav") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:6] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=477") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:7] Set("SIP/voipjet-0070afe0", "CDR(userfield)= 5083282553") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-0070afe0", "machinedetect") in new stack [Mar 14 18:49:50] -- Executing [s@liveonly:9] AMD("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:49:50] -- AMD: SIP/voipjet-0070afe0 6025161069 (null) (Fmt: 4) [Mar 14 18:49:50] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 18:49:52] -- AMD: Word detected. iWordsCount:1 [Mar 14 18:49:52] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 18:49:53] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 18:49:53] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-0070afe0", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 18:49:53] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-0070afe0", "0?liveonly-amd|s|1") in new stack [Mar 14 18:49:53] -- Executing [s@liveonly:12] Goto("SIP/voipjet-0070afe0", "liveonly-live|s|1") in new stack [Mar 14 18:49:53] -- Goto (liveonly-live,s,1) [Mar 14 18:49:53] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-0070afe0", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 18:49:53] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 18:49:58] == CDR updated on SIP/voipjet-0070afe0 [Mar 14 18:49:58] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-0070afe0", "Press 1 transfer") in new stack [Mar 14 18:49:58] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-0070afe0", "liveonly-transfer|s|1") in new stack [Mar 14 18:49:58] -- Goto (liveonly-transfer,s,1) [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "soundfile=20070314184958-1173912578.38.ulaw") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-0070afe0", "20070314184958-1173912578.38.ulaw|b") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CALLERID(num)=3024573281") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-0070afe0", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-0070afe0", "CDR(userfield5)=2007-03-14 18:49:58") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-0070afe0", "0?20") in new stack [Mar 14 18:49:58] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-0070afe0", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 18:49:58] RTP-stats [Mar 14 18:49:58] * Our Receiver: [Mar 14 18:49:58] == Begin MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:49:58] SSRC: 0 [Mar 14 18:49:58] Received packets: 0 [Mar 14 18:49:58] Lost packets: 0 [Mar 14 18:49:58] Jitter: 0.0000 [Mar 14 18:49:58] Transit: 0.0000 [Mar 14 18:49:58] RR-count: 0 [Mar 14 18:49:58] * Our Sender: [Mar 14 18:49:58] SSRC: 1959612917 [Mar 14 18:49:58] Sent packets: 0 [Mar 14 18:49:58] Lost packets: 0 [Mar 14 18:49:58] Jitter: 0 [Mar 14 18:49:58] SR-count: 0 [Mar 14 18:49:58] RTT: 0.000000 [Mar 14 18:49:58] Audio is at 167.206.178.7 port 11682 [Mar 14 18:49:58] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:49:58] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:49:58] 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=z9hG4bK6bca8a45;rport From: "3024573281" ;tag=as30cb6474 To: Contact: Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:49:58 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 11682 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 18:49:58] -- Called 13024573281@voipjet [Mar 14 18:49:58] -- Started music on hold, class 'trans3', on SIP/voipjet-0070afe0 [Mar 14 18:49:58] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6bca8a45;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as30cb6474 To: Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@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 18:49:58] --- (10 headers 0 lines) --- [Mar 14 18:50:00] <--- 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=z9hG4bK6bca8a45;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as30cb6474 To: ;tag=as761791cb Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@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 11164 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 18:50:00] --- (11 headers 10 lines) --- [Mar 14 18:50:00] Found RTP audio format 0 [Mar 14 18:50:00] Found RTP audio format 101 [Mar 14 18:50:00] Peer audio RTP is at port 69.16.233.35:11164 [Mar 14 18:50:00] Found description format PCMU for ID 0 [Mar 14 18:50:00] Found description format telephone-event for ID 101 [Mar 14 18:50:00] Got unsupported a:fmtp in SDP offer [Mar 14 18:50:00] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:50:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:50:00] Peer audio RTP is at port 69.16.233.35:11164 [Mar 14 18:50:00] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:50:00] -- Stopped music on hold on SIP/voipjet-0070afe0 [Mar 14 18:50:00] -- SIP/voipjet-006c0ff0 is making progress passing it to SIP/voipjet-0070afe0 [Mar 14 18:50:03] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK6bca8a45;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as30cb6474 To: ;tag=as761791cb Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@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 11164 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 18:50:03] --- (11 headers 10 lines) --- [Mar 14 18:50:03] Found RTP audio format 0 [Mar 14 18:50:03] Found RTP audio format 101 [Mar 14 18:50:03] Peer audio RTP is at port 69.16.233.35:11164 [Mar 14 18:50:03] Found description format PCMU for ID 0 [Mar 14 18:50:03] Found description format telephone-event for ID 101 [Mar 14 18:50:03] Got unsupported a:fmtp in SDP offer [Mar 14 18:50:03] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:50:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:50:03] Peer audio RTP is at port 69.16.233.35:11164 [Mar 14 18:50:03] list_route: hop: [Mar 14 18:50:03] set_destination: Parsing for address/port to send to [Mar 14 18:50:03] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:50:03] 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=z9hG4bK739bd7cf;rport From: "3024573281" ;tag=as30cb6474 To: ;tag=as761791cb Contact: Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:50:03] -- Call on SIP/voipjet-006c0ff0 left from hold [Mar 14 18:50:03] -- SIP/voipjet-006c0ff0 answered SIP/voipjet-0070afe0 [Mar 14 18:50:23] <--- 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=z9hG4bK7c0f7e95;rport From: ;tag=as5e37f444 To: "6025161069" ;tag=as49e679d2 Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 18:50:23] --- (9 headers 0 lines) --- [Mar 14 18:50:23] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 18:50:23] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK7c0f7e95;received=69.16.233.35;rport=5060 From: ;tag=as5e37f444 To: "6025161069" ;tag=as49e679d2 Call-ID: 4f245b9c09b8a41b01c7773c1df3c10d@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 18:50:23] Scheduling destruction of SIP dialog '47c67d7a0b119c950acabcd069fe4bcd@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 18:50:23] set_destination: Parsing for address/port to send to [Mar 14 18:50:23] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:50:23] 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=z9hG4bK48be42c3;rport From: "3024573281" ;tag=as30cb6474 To: ;tag=as761791cb Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:50:23] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-0070afe0' [Mar 14 18:50:23] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "CDR(userfield)=press1transfer 20070314184958-1173912578.38.ulaw") in new stack [Mar 14 18:50:23] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-0070afe0", "CDR(userfield2)=5083282553") in new stack [Mar 14 18:50:23] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 18:50:23] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=477") in new stack [Mar 14 18:50:23] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-0070afe0", "5083282553") in new stack [Mar 14 18:50:23] == End MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:50:23] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK48be42c3;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as30cb6474 To: ;tag=as761791cb Call-ID: 47c67d7a0b119c950acabcd069fe4bcd@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 18:50:23] --- (11 headers 0 lines) --- [Mar 14 18:50:23] Really destroying SIP dialog '47c67d7a0b119c950acabcd069fe4bcd@167.206.178.7' Method: INVITE [Mar 14 18:50:23] RTP-stats [Mar 14 18:50:23] * Our Receiver: [Mar 14 18:50:23] SSRC: 112342809 [Mar 14 18:50:23] Received packets: 1148 [Mar 14 18:50:23] Lost packets: 5 [Mar 14 18:50:23] Jitter: 0.0002 [Mar 14 18:50:23] Transit: -0.0004 [Mar 14 18:50:23] RR-count: 4 [Mar 14 18:50:23] * Our Sender: [Mar 14 18:50:23] SSRC: 1811825542 [Mar 14 18:50:23] Sent packets: 0 [Mar 14 18:50:23] Lost packets: 0 [Mar 14 18:50:23] Jitter: 0 [Mar 14 18:50:23] SR-count: 0 [Mar 14 18:50:23] RTT: 0.000000 [Mar 14 18:50:23] Really destroying SIP dialog '4f245b9c09b8a41b01c7773c1df3c10d@167.206.178.7' Method: BYE [Mar 14 18:50:23] RTP-stats [Mar 14 18:50:23] * Our Receiver: [Mar 14 18:50:23] SSRC: 762745693 [Mar 14 18:50:23] Received packets: 1935 [Mar 14 18:50:23] Lost packets: 9 [Mar 14 18:50:23] Jitter: 0.0002 [Mar 14 18:50:23] Transit: -0.0002 [Mar 14 18:50:23] RR-count: 1 [Mar 14 18:50:23] * Our Sender: [Mar 14 18:50:23] SSRC: 2020951459 [Mar 14 18:50:23] Sent packets: 1306 [Mar 14 18:50:23] Lost packets: 0 [Mar 14 18:50:23] Jitter: 0 [Mar 14 18:50:23] SR-count: 6 [Mar 14 18:50:23] RTT: 0.000000