ccccccccccccc (6102) is an X-Lite 6906 is on Cisco CCM 6 and is a Cisco 7960, running Cisco native protocol 192.168.10.10 is the Cisco Call Manager and also the MTP host 192.168.10.53 we assume to be the direct address for the Cisco 7960 192.168.130.140 is Asterisk 192.168.71.27 is the machine running X-Lite Call from X-Lite to Cisco. [2009-02-17 19:19:50.665] VERBOSE[452] logger.c: == Using SIP RTP CoS mark 5 [2009-02-17 19:19:50.867] VERBOSE[5468] logger.c: -- Executing [6906@AAAAAAAAAAAAAAAA:1] NoOp("SIP/ccccccccccccc-b75004e8", "") in new stack [2009-02-17 19:19:50.867] VERBOSE[5468] logger.c: -- Executing [6906@AAAAAAAAAAAAAAAA:2] Gosub("SIP/ccccccccccccc-b75004e8", "BBBBBBBBBBBBB(SIP/6906@192.168.10.10)") in new stack [2009-02-17 19:19:50.869] VERBOSE[5468] logger.c: -- Executing [6906@AAAAAAAAAAAAAAAA:51000] NoOp("SIP/ccccccccccccc-b75004e8", "") in new stack [2009-02-17 19:19:50.869] VERBOSE[5468] logger.c: -- Executing [6906@AAAAAAAAAAAAAAAA:51001] Dial("SIP/ccccccccccccc-b75004e8", "SIP/6906@192.168.10.10,600") in new stack [2009-02-17 19:19:50.870] VERBOSE[5468] logger.c: == Using SIP RTP CoS mark 5 [2009-02-17 19:19:50.889] VERBOSE[5468] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:19:50.889] VERBOSE[5468] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:19:50.889] VERBOSE[5468] logger.c: Adding codec 0x2 (gsm) to SDP [2009-02-17 19:19:50.889] VERBOSE[5468] logger.c: Adding codec 0x8 (alaw) to SDP [2009-02-17 19:19:50.890] VERBOSE[5468] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:19:50.890] VERBOSE[5468] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: INVITE sip:6906@192.168.10.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5c7a216d;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0 Date: Tue, 17 Feb 2009 19:19:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 287 v=0 o=root 1710661285 1710661285 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 10844 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2009-02-17 19:19:50.891] VERBOSE[5468] logger.c: -- Called 6906@192.168.10.10 [2009-02-17 19:19:50.907] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 100 Trying Date: Tue, 17 Feb 2009 18:51:57 GMT From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence Content-Length: 0 To: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5c7a216d;rport CSeq: 102 INVITE <-------------> [2009-02-17 19:19:50.907] VERBOSE[452] logger.c: --- (9 headers 0 lines) --- [2009-02-17 19:19:50.922] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 180 Ringing Date: Tue, 17 Feb 2009 18:51:57 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence Remote-Party-ID: ;party=called;screen=yes;privacy=off Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5c7a216d;rport CSeq: 102 INVITE <-------------> [2009-02-17 19:19:50.922] VERBOSE[452] logger.c: --- (12 headers 0 lines) --- [2009-02-17 19:19:50.922] VERBOSE[5468] logger.c: -- SIP/192.168.10.10-09e2f330 is ringing [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Tue, 17 Feb 2009 18:51:57 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence, kpml Supported: replaces Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=called;screen=yes;privacy=off Content-Length: 214 Require: timer To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5c7a216d;rport CSeq: 102 INVITE Session-Expires: 1800;refresher=uas v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.53 t=0 0 m=audio 17954 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: --- (16 headers 10 lines) --- [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:17954 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:17954 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: list_route: hop: [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 19:19:53.174] VERBOSE[452] logger.c: Transmitting (no NAT) to 192.168.10.10:5060: ACK sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK3e0b5df4;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0 Content-Length: 0 --- [2009-02-17 19:19:53.174] VERBOSE[5468] logger.c: -- SIP/192.168.10.10-09e2f330 answered SIP/ccccccccccccc-b75004e8 [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: -- Native bridging SIP/ccccccccccccc-b75004e8 and SIP/192.168.10.10-09e2f330 [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:19:53.207] VERBOSE[5468] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: INVITE sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK53d082f4;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 236 v=0 o=root 1710661285 1710661286 IN IP4 192.168.71.27 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.71.27 t=0 0 m=audio 41506 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2009-02-17 19:19:53.209] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 100 Trying Date: Tue, 17 Feb 2009 18:52:00 GMT From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence, kpml Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK53d082f4;rport CSeq: 103 INVITE <-------------> [2009-02-17 19:19:53.209] VERBOSE[452] logger.c: --- (9 headers 0 lines) --- [2009-02-17 19:19:53.210] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Tue, 17 Feb 2009 18:52:00 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence, kpml Supported: replaces Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=called;screen=yes;privacy=off Content-Length: 214 Require: timer To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK53d082f4;rport CSeq: 103 INVITE Session-Expires: 1800;refresher=uas v=0 o=CiscoSystemsCCM-SIP 2000 2 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.53 t=0 0 m=audio 17954 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:19:53.210] VERBOSE[452] logger.c: --- (16 headers 10 lines) --- [2009-02-17 19:19:53.210] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:19:53.210] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:19:53.210] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:17954 [2009-02-17 19:19:53.211] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:19:53.211] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:19:53.211] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:19:53.211] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:19:53.211] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:17954 [2009-02-17 19:19:53.212] VERBOSE[452] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 19:19:53.212] VERBOSE[452] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 19:19:53.212] VERBOSE[452] logger.c: Transmitting (no NAT) to 192.168.10.10:5060: ACK sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK3a07e3fc;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0 Content-Length: 0 --- [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: ******************** Initiate hold on Cisco ********************** <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 18:52:08 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 197921280-2578515789-2464855-168470720 Content-Length: 220 User-Agent: Cisco-CUCM6.1 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Contact: Expires: 180 Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1734a6948d CSeq: 101 INVITE Session-Expires: 1800;refresher=uac Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 3 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 17954 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: --- (20 headers 11 lines) --- [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Peer audio RTP is at port 0.0.0.0:17954 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: Peer audio RTP is at port 0.0.0.0:17954 [2009-02-17 19:20:00.999] VERBOSE[452] logger.c: <--- Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1734a6948d;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-02-17 19:20:01.000] VERBOSE[452] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:20:01.000] VERBOSE[452] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:20:01.000] VERBOSE[452] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:20:01.000] VERBOSE[452] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1734a6948d;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 1710661285 1710661287 IN IP4 192.168.71.27 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.71.27 t=0 0 m=audio 41506 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=inactive <------------> [2009-02-17 19:20:01.003] VERBOSE[5468] logger.c: -- Started music on hold, class 'default', on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:20:01.004] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 18:52:08 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Content-Length: 0 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1820320682 CSeq: 101 ACK Max-Forwards: 70 <-------------> [2009-02-17 19:20:01.005] VERBOSE[452] logger.c: --- (10 headers 0 lines) --- [2009-02-17 19:20:01.005] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 18:52:08 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 197921280-2578515789-2464855-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Contact: Expires: 180 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1928f10129 CSeq: 102 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-02-17 19:20:01.006] VERBOSE[452] logger.c: --- (19 headers 0 lines) --- [2009-02-17 19:20:01.006] VERBOSE[452] logger.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-02-17 19:20:01.007] VERBOSE[452] logger.c: <--- Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1928f10129;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-02-17 19:20:01.008] VERBOSE[452] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:20:01.009] VERBOSE[452] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:20:01.010] VERBOSE[452] logger.c: Adding codec 0x2 (gsm) to SDP [2009-02-17 19:20:01.010] VERBOSE[452] logger.c: Adding codec 0x8 (alaw) to SDP [2009-02-17 19:20:01.010] VERBOSE[452] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:20:01.012] VERBOSE[452] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1928f10129;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 1710661285 1710661288 IN IP4 192.168.71.27 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.71.27 t=0 0 m=audio 41506 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=inactive <------------> [2009-02-17 19:20:01.028] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 18:52:08 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Content-Length: 225 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1a1a75f0fd CSeq: 102 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 4 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.10 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:20:01.028] VERBOSE[452] logger.c: --- (11 headers 11 lines) --- [2009-02-17 19:20:01.028] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:20:01.028] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:20:01.029] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 19:20:01.029] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:20:01.029] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:20:01.030] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:20:01.030] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:20:01.031] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 19:20:01.091] VERBOSE[5468] logger.c: -- Stopped music on hold on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:20:01.091] VERBOSE[5468] logger.c: -- Started music on hold, class 'default', on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:21:03.162] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:24:03.440] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:27:03.676] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:30:03.964] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:33:04.168] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: ********************************** Resume Cisco Phone ********************* <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 19:05:43 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 197921280-2578515789-2464855-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Contact: Expires: 180 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1b18fd7855 CSeq: 103 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: --- (19 headers 0 lines) --- [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: <--- Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1b18fd7855;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Adding codec 0x2 (gsm) to SDP [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Adding codec 0x8 (alaw) to SDP [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:33:36.385] VERBOSE[452] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1b18fd7855;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 1710661285 1710661289 IN IP4 192.168.71.27 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.71.27 t=0 0 m=audio 41506 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=inactive <------------> [2009-02-17 19:33:36.391] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 19:05:43 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Content-Length: 225 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1c6350f148 CSeq: 103 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 6 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.53 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:33:36.391] VERBOSE[452] logger.c: --- (11 headers 11 lines) --- [2009-02-17 19:33:36.393] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:33:36.394] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:33:36.398] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-02-17 19:33:36.399] VERBOSE[5468] logger.c: -- Stopped music on hold on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:33:36.399] VERBOSE[5468] logger.c: -- Started music on hold, class 'default', on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:36:04.414] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc ********************************* Hold Cisco again ************************ [2009-02-17 19:36:42.521] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 19:08:49 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 197921280-2578515789-2464855-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Contact: Expires: 180 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1d139dff60 CSeq: 104 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-02-17 19:36:42.521] VERBOSE[452] logger.c: --- (19 headers 0 lines) --- [2009-02-17 19:36:42.521] VERBOSE[452] logger.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: <--- Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1d139dff60;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: Adding codec 0x2 (gsm) to SDP [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: Adding codec 0x8 (alaw) to SDP [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 19:36:42.522] VERBOSE[452] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1d139dff60;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 283 v=0 o=root 1710661285 1710661290 IN IP4 192.168.71.27 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.71.27 t=0 0 m=audio 41506 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=inactive <------------> [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6102@192.168.130.140:5060 SIP/2.0 Date: Tue, 17 Feb 2009 19:08:49 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Allow-Events: presence, kpml Content-Length: 225 To: "DDDDDDDDDDDDD" ;tag=as064c1570 Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2f1e35540069 CSeq: 104 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 8 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.10 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: --- (11 headers 11 lines) --- [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 19:36:42.526] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 19:36:42.527] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 19:36:42.527] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 19:36:42.527] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 19:36:42.527] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 19:36:42.527] VERBOSE[5468] logger.c: -- Stopped music on hold on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:36:42.527] VERBOSE[5468] logger.c: -- Started music on hold, class 'default', on SIP/ccccccccccccc-b75004e8 [2009-02-17 19:39:04.620] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:42:04.827] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:45:05.059] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:47:37.498] VERBOSE[5452] logger.c: -- Remote UNIX connection disconnected [2009-02-17 19:48:05.338] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:51:05.545] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:54:05.760] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 19:57:05.964] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc [2009-02-17 20:00:06.172] NOTICE[452] chan_sip.c: Received SIP subscribe for peer without mailbox: ccccccccccccc ******************** Machine running X-Lite shut down ***************** [2009-02-17 20:02:40.163] VERBOSE[5468] logger.c: -- Stopped music on hold on SIP/ccccccccccccc-b75004e8 [2009-02-17 20:02:40.165] VERBOSE[5468] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 20:02:40.165] VERBOSE[5468] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 20:02:40.167] VERBOSE[5468] logger.c: Audio is at 192.168.130.140 port 10844 [2009-02-17 20:02:40.168] VERBOSE[5468] logger.c: Adding codec 0x4 (ulaw) to SDP [2009-02-17 20:02:40.170] VERBOSE[5468] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-02-17 20:02:40.171] VERBOSE[5468] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: INVITE sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK77bdb9f2;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.0 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 1710661285 1710661291 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 10844 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=inactive --- [2009-02-17 20:02:40.178] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 100 Trying Date: Tue, 17 Feb 2009 19:34:46 GMT From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence, kpml Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK77bdb9f2;rport CSeq: 104 INVITE <-------------> [2009-02-17 20:02:40.178] VERBOSE[452] logger.c: --- (9 headers 0 lines) --- [2009-02-17 20:02:40.179] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Tue, 17 Feb 2009 19:34:46 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "DDDDDDDDDDDDD" ;tag=as064c1570 Allow-Events: presence, kpml Supported: replaces Remote-Party-ID: "EEEEEEEEEEEE 6906" ;party=called;screen=yes;privacy=off Content-Length: 225 Require: timer To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Content-Type: application/sdp Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK77bdb9f2;rport CSeq: 104 INVITE Session-Expires: 1800;refresher=uas v=0 o=CiscoSystemsCCM-SIP 2000 9 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.10 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-02-17 20:02:40.179] VERBOSE[452] logger.c: --- (16 headers 11 lines) --- [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Found RTP audio format 0 [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Found RTP audio format 101 [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Found audio description format PCMU for ID 0 [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Found audio description format telephone-event for ID 101 [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-02-17 20:02:40.182] VERBOSE[452] logger.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-02-17 20:02:40.183] VERBOSE[452] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 20:02:40.183] VERBOSE[452] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 20:02:40.184] VERBOSE[452] logger.c: Transmitting (no NAT) to 192.168.10.10:5060: ACK sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK206d0893;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Contact: Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.0 Content-Length: 0 --- [2009-02-17 20:02:40.189] VERBOSE[5468] logger.c: Scheduling destruction of SIP dialog '0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140' in 32000 ms (Method: ACK) [2009-02-17 20:02:40.191] VERBOSE[5468] logger.c: set_destination: Parsing for address/port to send to [2009-02-17 20:02:40.191] VERBOSE[5468] logger.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-02-17 20:02:40.192] VERBOSE[5468] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: BYE sip:6906@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK79cbf08a;rport Max-Forwards: 70 From: "DDDDDDDDDDDDD" ;tag=as064c1570 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 CSeq: 105 BYE User-Agent: Asterisk PBX 1.6.0 Content-Length: 0 --- [2009-02-17 20:02:40.197] VERBOSE[452] logger.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Tue, 17 Feb 2009 19:34:46 GMT From: "DDDDDDDDDDDDD" ;tag=as064c1570 Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-25977509 Call-ID: 0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK79cbf08a;rport CSeq: 105 BYE <-------------> [2009-02-17 20:02:40.197] VERBOSE[452] logger.c: --- (8 headers 0 lines) --- [2009-02-17 20:02:40.198] VERBOSE[452] logger.c: Really destroying SIP dialog '0b74c5fb09e0f1d2675e13642f812a44@192.168.130.140' Method: ACK [2009-02-17 20:02:40.210] VERBOSE[5468] logger.c: == Spawn extension (AAAAAAAAAAAAAAAA, 6906, 51001) exited non-zero on 'SIP/ccccccccccccc-b75004e8' [2009-02-17 20:02:40.228] VERBOSE[452] logger.c: -- Unregistered SIP 'ccccccccccccc'