ASTEISK05*CLI> [Mar 14 18:36:56] RTP-stats [Mar 14 18:36:56] * Our Receiver: [Mar 14 18:36:56] SSRC: 0 [Mar 14 18:36:56] Received packets: 0 [Mar 14 18:36:56] Lost packets: 0 [Mar 14 18:36:56] Jitter: 0.0000 [Mar 14 18:36:56] Transit: 0.0000 [Mar 14 18:36:56] RR-count: 0 [Mar 14 18:36:56] * Our Sender: [Mar 14 18:36:56] SSRC: 249920996 [Mar 14 18:36:56] Sent packets: 0 [Mar 14 18:36:56] Lost packets: 0 [Mar 14 18:36:56] Jitter: 0 [Mar 14 18:36:56] SR-count: 0 [Mar 14 18:36:56] RTT: 0.000000 [Mar 14 18:36:56] Audio is at 167.206.178.7 port 19812 [Mar 14 18:36:56] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:36:56] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:36:56] 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=z9hG4bK07867abc;rport From: "6025161069" ;tag=as7925934e To: Contact: Call-ID: 0a07ce603f316ca026a852b817b6d653@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:36:56 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 19812 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 18:36:56] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK07867abc;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7925934e To: Call-ID: 0a07ce603f316ca026a852b817b6d653@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 18:36:56] --- (10 headers 0 lines) --- [Mar 14 18:37:02] <--- 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=z9hG4bK07867abc;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7925934e To: ;tag=as077e4e00 Call-ID: 0a07ce603f316ca026a852b817b6d653@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 10008 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 18:37:02] --- (11 headers 10 lines) --- [Mar 14 18:37:02] Found RTP audio format 0 [Mar 14 18:37:02] Found RTP audio format 101 [Mar 14 18:37:02] Peer audio RTP is at port 69.16.233.35:10008 [Mar 14 18:37:02] Found description format PCMU for ID 0 [Mar 14 18:37:02] Found description format telephone-event for ID 101 [Mar 14 18:37:02] Got unsupported a:fmtp in SDP offer [Mar 14 18:37:02] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:37:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:37:02] Peer audio RTP is at port 69.16.233.35:10008 [Mar 14 18:37:07] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK07867abc;received=167.206.178.7;rport=5060 From: "6025161069" ;tag=as7925934e To: ;tag=as077e4e00 Call-ID: 0a07ce603f316ca026a852b817b6d653@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 10008 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 18:37:07] --- (11 headers 10 lines) --- [Mar 14 18:37:07] Found RTP audio format 0 [Mar 14 18:37:07] Found RTP audio format 101 [Mar 14 18:37:07] Peer audio RTP is at port 69.16.233.35:10008 [Mar 14 18:37:07] Found description format PCMU for ID 0 [Mar 14 18:37:07] Found description format telephone-event for ID 101 [Mar 14 18:37:07] Got unsupported a:fmtp in SDP offer [Mar 14 18:37:07] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:37:07] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:37:07] Peer audio RTP is at port 69.16.233.35:10008 [Mar 14 18:37:07] list_route: hop: [Mar 14 18:37:07] set_destination: Parsing for address/port to send to [Mar 14 18:37:07] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:37:07] 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=z9hG4bK2d418e6f;rport From: "6025161069" ;tag=as7925934e To: ;tag=as077e4e00 Contact: Call-ID: 0a07ce603f316ca026a852b817b6d653@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:37:07] > Channel SIP/voipjet-0070afe0 was answered. [Mar 14 18:37:07] -- Executing [s@liveonly:1] Answer("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:37:07] -- Executing [s@liveonly:2] NoOp("SIP/voipjet-0070afe0", "liveonly message:/var/lib/asterisk/sounds/sound26 M2: M3:3 DID:3024573281 phonenumber:5083282553 Campaign ID:471 CALLERID: ") in new stack [Mar 14 18:37:07] -- Executing [s@liveonly:3] Wait("SIP/voipjet-0070afe0", "1") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:4] Set("SIP/voipjet-0070afe0", "soundfile=DEBUG-20070314183708-1173911816.26.wav") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:5] NoOp("SIP/voipjet-0070afe0", "DEBUG-20070314183708-1173911816.26.wav") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:6] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=471") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:7] Set("SIP/voipjet-0070afe0", "CDR(userfield)= 5083282553") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:8] NoOp("SIP/voipjet-0070afe0", "machinedetect") in new stack [Mar 14 18:37:08] -- Executing [s@liveonly:9] AMD("SIP/voipjet-0070afe0", "") in new stack [Mar 14 18:37:08] -- AMD: SIP/voipjet-0070afe0 6025161069 (null) (Fmt: 4) [Mar 14 18:37:08] -- AMD: initialSilence [5700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [860] [Mar 14 18:37:09] -- AMD: Word detected. iWordsCount:1 [Mar 14 18:37:09] -- AMD: Changed state to STATE_IN_SILENCE [Mar 14 18:37:10] -- AMD: HUMAN: silenceDuration:1200 afterGreetingSilence:1200 [Mar 14 18:37:10] -- Executing [s@liveonly:10] NoOp("SIP/voipjet-0070afe0", "AMD got: HUMAN-1200-1200") in new stack [Mar 14 18:37:10] -- Executing [s@liveonly:11] GotoIf("SIP/voipjet-0070afe0", "0?liveonly-amd|s|1") in new stack [Mar 14 18:37:10] -- Executing [s@liveonly:12] Goto("SIP/voipjet-0070afe0", "liveonly-live|s|1") in new stack [Mar 14 18:37:10] -- Goto (liveonly-live,s,1) [Mar 14 18:37:10] -- Executing [s@liveonly-live:1] BackGround("SIP/voipjet-0070afe0", "/var/lib/asterisk/sounds/sound26") in new stack [Mar 14 18:37:10] -- Playing '/var/lib/asterisk/sounds/sound26' (language 'en') [Mar 14 18:37:21] == CDR updated on SIP/voipjet-0070afe0 [Mar 14 18:37:21] -- Executing [1@liveonly-live:1] NoOp("SIP/voipjet-0070afe0", "Press 1 transfer") in new stack [Mar 14 18:37:21] -- Executing [1@liveonly-live:2] Goto("SIP/voipjet-0070afe0", "liveonly-transfer|s|1") in new stack [Mar 14 18:37:21] -- Goto (liveonly-transfer,s,1) [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "soundfile=20070314183721-1173911816.26.ulaw") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:2] MixMonitor("SIP/voipjet-0070afe0", "20070314183721-1173911816.26.ulaw|b") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CALLERID(num)=3024573281") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:4] ExecIf("SIP/voipjet-0070afe0", "0|Set|CALLERID(num)=5083282553") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:5] Set("SIP/voipjet-0070afe0", "CDR(userfield5)=2007-03-14 18:37:21") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:6] GotoIf("SIP/voipjet-0070afe0", "0?20") in new stack [Mar 14 18:37:21] -- Executing [s@liveonly-transfer:7] Dial("SIP/voipjet-0070afe0", "SIP/13024573281@voipjet||ojm(trans3)") in new stack [Mar 14 18:37:21] RTP-stats [Mar 14 18:37:21] * Our Receiver: [Mar 14 18:37:21] == Begin MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:37:21] SSRC: 0 [Mar 14 18:37:21] Received packets: 0 [Mar 14 18:37:21] Lost packets: 0 [Mar 14 18:37:21] Jitter: 0.0000 [Mar 14 18:37:21] Transit: 0.0000 [Mar 14 18:37:21] RR-count: 0 [Mar 14 18:37:21] * Our Sender: [Mar 14 18:37:21] SSRC: 1230751012 [Mar 14 18:37:21] Sent packets: 0 [Mar 14 18:37:21] Lost packets: 0 [Mar 14 18:37:21] Jitter: 0 [Mar 14 18:37:21] SR-count: 0 [Mar 14 18:37:21] RTT: 0.000000 [Mar 14 18:37:21] Audio is at 167.206.178.7 port 19476 [Mar 14 18:37:21] Adding codec 0x4 (ulaw) to SDP [Mar 14 18:37:21] Adding non-codec 0x1 (telephone-event) to SDP [Mar 14 18:37:21] 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=z9hG4bK529e155c;rport From: "3024573281" ;tag=as0952ff0d To: Contact: Call-ID: 3ee6634a00ef27f225152963612b803d@167.206.178.7 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 22:37:21 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 19476 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 18:37:21] -- Called 13024573281@voipjet [Mar 14 18:37:21] -- Started music on hold, class 'trans3', on SIP/voipjet-0070afe0 [Mar 14 18:37:21] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK529e155c;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as0952ff0d To: Call-ID: 3ee6634a00ef27f225152963612b803d@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 18:37:21] --- (10 headers 0 lines) --- [Mar 14 18:37:24] <--- 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=z9hG4bK529e155c;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as0952ff0d To: ;tag=as7d5015da Call-ID: 3ee6634a00ef27f225152963612b803d@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 14874 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 18:37:24] --- (11 headers 10 lines) --- [Mar 14 18:37:24] Found RTP audio format 0 [Mar 14 18:37:24] Found RTP audio format 101 [Mar 14 18:37:24] Peer audio RTP is at port 69.16.233.35:14874 [Mar 14 18:37:24] Found description format PCMU for ID 0 [Mar 14 18:37:24] Found description format telephone-event for ID 101 [Mar 14 18:37:24] Got unsupported a:fmtp in SDP offer [Mar 14 18:37:24] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:37:24] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:37:24] Peer audio RTP is at port 69.16.233.35:14874 [Mar 14 18:37:24] -- Call on SIP/voipjet-ae0285e0 left from hold [Mar 14 18:37:24] -- Stopped music on hold on SIP/voipjet-0070afe0 [Mar 14 18:37:24] -- SIP/voipjet-ae0285e0 is making progress passing it to SIP/voipjet-0070afe0 [Mar 14 18:37:27] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK529e155c;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as0952ff0d To: ;tag=as7d5015da Call-ID: 3ee6634a00ef27f225152963612b803d@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 14874 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 18:37:27] --- (11 headers 10 lines) --- [Mar 14 18:37:27] Found RTP audio format 0 [Mar 14 18:37:27] Found RTP audio format 101 [Mar 14 18:37:27] Peer audio RTP is at port 69.16.233.35:14874 [Mar 14 18:37:27] Found description format PCMU for ID 0 [Mar 14 18:37:27] Found description format telephone-event for ID 101 [Mar 14 18:37:27] Got unsupported a:fmtp in SDP offer [Mar 14 18:37:27] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 14 18:37:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 14 18:37:27] Peer audio RTP is at port 69.16.233.35:14874 [Mar 14 18:37:27] list_route: hop: [Mar 14 18:37:27] set_destination: Parsing for address/port to send to [Mar 14 18:37:27] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:37:27] 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=z9hG4bK039030c9;rport From: "3024573281" ;tag=as0952ff0d To: ;tag=as7d5015da Contact: Call-ID: 3ee6634a00ef27f225152963612b803d@167.206.178.7 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:37:27] -- Call on SIP/voipjet-ae0285e0 left from hold [Mar 14 18:37:27] -- SIP/voipjet-ae0285e0 answered SIP/voipjet-0070afe0 [Mar 14 18:37:40] <--- 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=z9hG4bK6fe97240;rport From: ;tag=as077e4e00 To: "6025161069" ;tag=as7925934e Call-ID: 0a07ce603f316ca026a852b817b6d653@167.206.178.7 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Mar 14 18:37:40] --- (9 headers 0 lines) --- [Mar 14 18:37:40] Sending to 69.16.233.35 : 5060 (NAT) [Mar 14 18:37:40] <--- Transmitting (NAT) to 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.16.233.35:5060;branch=z9hG4bK6fe97240;received=69.16.233.35;rport=5060 From: ;tag=as077e4e00 To: "6025161069" ;tag=as7925934e Call-ID: 0a07ce603f316ca026a852b817b6d653@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 18:37:40] Scheduling destruction of SIP dialog '3ee6634a00ef27f225152963612b803d@167.206.178.7' in 6400 ms (Method: INVITE) [Mar 14 18:37:40] set_destination: Parsing for address/port to send to [Mar 14 18:37:40] set_destination: set destination to 69.16.233.35, port 5060 [Mar 14 18:37:40] 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=z9hG4bK07c7c442;rport From: "3024573281" ;tag=as0952ff0d To: ;tag=as7d5015da Call-ID: 3ee6634a00ef27f225152963612b803d@167.206.178.7 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 14 18:37:40] == Spawn extension (liveonly-transfer, s, 7) exited non-zero on 'SIP/voipjet-0070afe0' [Mar 14 18:37:40] -- Executing [h@liveonly-transfer:1] Set("SIP/voipjet-0070afe0", "CDR(userfield)=press1transfer 20070314183721-1173911816.26.ulaw") in new stack [Mar 14 18:37:40] -- Executing [h@liveonly-transfer:2] Set("SIP/voipjet-0070afe0", "CDR(userfield2)=5083282553") in new stack [Mar 14 18:37:40] -- Executing [h@liveonly-transfer:3] Set("SIP/voipjet-0070afe0", "CDR(userfield3)=HUMAN-1200-1200") in new stack [Mar 14 18:37:40] -- Executing [h@liveonly-transfer:4] Set("SIP/voipjet-0070afe0", "CDR(accountcode)=471") in new stack [Mar 14 18:37:40] -- Executing [h@liveonly-transfer:5] NoOp("SIP/voipjet-0070afe0", "5083282553") in new stack [Mar 14 18:37:40] == End MixMonitor Recording SIP/voipjet-0070afe0 [Mar 14 18:37:40] <--- SIP read from 69.16.233.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 167.206.178.7:5060;branch=z9hG4bK07c7c442;received=167.206.178.7;rport=5060 From: "3024573281" ;tag=as0952ff0d To: ;tag=as7d5015da Call-ID: 3ee6634a00ef27f225152963612b803d@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 18:37:40] --- (11 headers 0 lines) --- [Mar 14 18:37:40] Really destroying SIP dialog '3ee6634a00ef27f225152963612b803d@167.206.178.7' Method: INVITE [Mar 14 18:37:40] RTP-stats [Mar 14 18:37:40] * Our Receiver: [Mar 14 18:37:40] SSRC: 631349829 [Mar 14 18:37:40] Received packets: 824 [Mar 14 18:37:40] Lost packets: 1 [Mar 14 18:37:40] Jitter: 0.0005 [Mar 14 18:37:40] Transit: -0.0001 [Mar 14 18:37:40] RR-count: 0 [Mar 14 18:37:40] * Our Sender: [Mar 14 18:37:40] SSRC: 701701781 [Mar 14 18:37:40] Sent packets: 353 [Mar 14 18:37:40] Lost packets: 0 [Mar 14 18:37:40] Jitter: 0 [Mar 14 18:37:40] SR-count: 3 [Mar 14 18:37:40] RTT: 0.000000 [Mar 14 18:37:40] Really destroying SIP dialog '0a07ce603f316ca026a852b817b6d653@167.206.178.7' Method: BYE [Mar 14 18:37:40] RTP-stats [Mar 14 18:37:40] * Our Receiver: [Mar 14 18:37:40] SSRC: 329248068 [Mar 14 18:37:40] Received packets: 1580 [Mar 14 18:37:40] Lost packets: 1 [Mar 14 18:37:40] Jitter: 0.0002 [Mar 14 18:37:40] Transit: 0.0000 [Mar 14 18:37:40] RR-count: 1 [Mar 14 18:37:40] * Our Sender: [Mar 14 18:37:40] SSRC: 1521986165 [Mar 14 18:37:40] Sent packets: 1352 [Mar 14 18:37:40] Lost packets: 0 [Mar 14 18:37:40] Jitter: 0 [Mar 14 18:37:40] SR-count: 6 [Mar 14 18:37:40] RTT: 0.000000