[2009-07-02 16:08:49.317] VERBOSE[10893] pbx.c: -- Executing [6906@aaaaaaaaaaaaaaaa:52005] Dial("SIP/6105-09d6b5e8", "SIP/6906@cisco1,600") in new stack [2009-07-02 16:08:49.320] VERBOSE[10893] netsock.c: == Using SIP RTP CoS mark 5 [2009-07-02 16:08:49.325] VERBOSE[10893] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:08:49.326] VERBOSE[10893] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:08:49.326] VERBOSE[10893] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:08:49.328] VERBOSE[10893] chan_sip.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=z9hG4bK55e67319;rport Max-Forwards: 70 From: "6105" ;tag=as74b09d4c To: Contact: Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 02 Jul 2009 15:08:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 240 v=0 o=root 555307107 555307107 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 --- [2009-07-02 16:08:49.329] VERBOSE[10893] app_dial.c: -- Called 6906@cisco1 [2009-07-02 16:08:49.331] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 100 Trying Date: Thu, 02 Jul 2009 15:01:25 GMT From: "6105" ;tag=as74b09d4c Allow-Events: presence Content-Length: 0 To: Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK55e67319;rport CSeq: 102 INVITE <-------------> [2009-07-02 16:08:49.331] VERBOSE[10140] chan_sip.c: --- (9 headers 0 lines) --- [2009-07-02 16:08:49.380] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 180 Ringing Date: Thu, 02 Jul 2009 15:01:25 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "6105" ;tag=as74b09d4c Allow-Events: presence Remote-Party-ID: ;party=called;screen=yes;privacy=off Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Contact: Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK55e67319;rport CSeq: 102 INVITE <-------------> [2009-07-02 16:08:49.380] VERBOSE[10140] chan_sip.c: --- (12 headers 0 lines) --- [2009-07-02 16:08:49.383] VERBOSE[10893] app_dial.c: -- SIP/cisco1-09d71568 is ringing [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:01:25 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: "6105" ;tag=as74b09d4c Allow-Events: presence, kpml Supported: replaces Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=called;screen=yes;privacy=off Content-Length: 214 Require: timer To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Contact: Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK55e67319;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 29872 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: --- (16 headers 10 lines) --- [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:29872 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:29872 [2009-07-02 16:08:54.325] VERBOSE[10140] chan_sip.c: list_route: hop: [2009-07-02 16:08:54.335] VERBOSE[10140] chan_sip.c: set_destination: Parsing for address/port to send to [2009-07-02 16:08:54.335] VERBOSE[10140] chan_sip.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-07-02 16:08:54.335] VERBOSE[10140] chan_sip.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=z9hG4bK39ddef55;rport Max-Forwards: 70 From: "6105" ;tag=as74b09d4c To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Contact: Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0 Content-Length: 0 --- [2009-07-02 16:08:54.339] VERBOSE[10893] app_dial.c: -- SIP/cisco1-09d71568 answered SIP/6105-09d6b5e8 [2009-07-02 16:08:54.350] VERBOSE[10893] rtp.c: -- Packet2Packet bridging SIP/6105-09d6b5e8 and SIP/cisco1-09d71568 [2009-07-02 16:08:58.589] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:34 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 220 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d4498f5c19 CSeq: 101 INVITE Session-Expires: 1800;refresher=uac Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 2 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 29872 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 <-------------> [2009-07-02 16:08:58.589] VERBOSE[10140] chan_sip.c: --- (20 headers 11 lines) --- [2009-07-02 16:08:58.589] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:08:58.589] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:08:58.590] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:08:58.590] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 0.0.0.0:29872 [2009-07-02 16:08:58.591] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:08:58.591] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:08:58.591] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:08:58.591] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:08:58.591] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 0.0.0.0:29872 [2009-07-02 16:08:58.592] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d4498f5c19;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:08:58.593] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:08:58.593] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:08:58.593] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:08:58.594] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d4498f5c19;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307108 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:08:58.597] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:08:58.599] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:34 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 0 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d51d52dff CSeq: 101 ACK Max-Forwards: 70 <-------------> [2009-07-02 16:08:58.600] VERBOSE[10140] chan_sip.c: --- (10 headers 0 lines) --- [2009-07-02 16:08:58.600] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:34 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d62d70ba9a CSeq: 102 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:08:58.600] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:08:58.601] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:08:58.602] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d62d70ba9a;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:08:58.603] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:08:58.604] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:08:58.604] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:08:58.604] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d62d70ba9a;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307109 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:08:58.622] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:34 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 225 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d72e73aa0 CSeq: 102 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 3 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 <-------------> [2009-07-02 16:08:58.622] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:08:58.622] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:08:58.622] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:08:58.623] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:08:58.634] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:08:58.635] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:09:03.066] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:39 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d81152ae57 CSeq: 103 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:09:03.066] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d81152ae57;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 103 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:09:03.067] VERBOSE[10140] chan_sip.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=z9hG4bK5f6d81152ae57;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 103 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307110 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:39 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 225 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6d92bb1bf9e CSeq: 103 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 5 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 <-------------> [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:09:03.080] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:03.081] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:09:03.081] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:43 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6da23e8bd5 CSeq: 104 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.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=z9hG4bK5f6da23e8bd5;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 104 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:09:06.949] VERBOSE[10140] chan_sip.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=z9hG4bK5f6da23e8bd5;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 104 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307111 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:09:06.953] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:43 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 225 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6db63abd960 CSeq: 104 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 7 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 <-------------> [2009-07-02 16:09:06.953] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:09:06.953] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:09:06.953] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:09:06.954] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:09:06.954] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:09:06.954] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:09:08.600] VERBOSE[10140] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: OPTIONS sip:192.168.10.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK68a02ab3;rport Max-Forwards: 70 From: "asterisk" ;tag=as7e05e758 To: Contact: Call-ID: 0a09b64228c9b47408f8de951ab0543d@192.168.130.140 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 02 Jul 2009 15:09:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [2009-07-02 16:09:08.602] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:01:44 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS From: "asterisk" ;tag=as7e05e758 Content-Length: 0 To: ;tag=1802345136 Call-ID: 0a09b64228c9b47408f8de951ab0543d@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK68a02ab3;rport CSeq: 102 OPTIONS <-------------> [2009-07-02 16:09:08.602] VERBOSE[10140] chan_sip.c: --- (9 headers 0 lines) --- [2009-07-02 16:09:08.602] VERBOSE[10140] chan_sip.c: Really destroying SIP dialog '0a09b64228c9b47408f8de951ab0543d@192.168.130.140' Method: OPTIONS [2009-07-02 16:09:08.679] VERBOSE[10140] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: OPTIONS sip:192.168.10.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK7c429548;rport Max-Forwards: 70 From: "asterisk" ;tag=as1af9212d To: Contact: Call-ID: 5df5f1bd47972f464ed9bf7b3b7b79b0@192.168.130.140 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 02 Jul 2009 15:09:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [2009-07-02 16:09:08.681] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:01:44 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS From: "asterisk" ;tag=as1af9212d Content-Length: 0 To: ;tag=1930920128 Call-ID: 5df5f1bd47972f464ed9bf7b3b7b79b0@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK7c429548;rport CSeq: 102 OPTIONS <-------------> [2009-07-02 16:09:08.681] VERBOSE[10140] chan_sip.c: --- (9 headers 0 lines) --- [2009-07-02 16:09:08.681] VERBOSE[10140] chan_sip.c: Really destroying SIP dialog '5df5f1bd47972f464ed9bf7b3b7b79b0@192.168.130.140' Method: OPTIONS [2009-07-02 16:09:10.050] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:46 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6dc2fc8fabf CSeq: 105 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:09:10.050] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.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=z9hG4bK5f6dc2fc8fabf;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 105 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:09:10.051] VERBOSE[10140] chan_sip.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=z9hG4bK5f6dc2fc8fabf;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 105 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307112 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:46 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 225 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6dd72891a13 CSeq: 105 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 9 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 <-------------> [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:09:10.056] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:10.057] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:09:10.057] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:48 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6de3da5aeaa CSeq: 106 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.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=z9hG4bK5f6de3da5aeaa;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 106 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:09:12.513] VERBOSE[10140] chan_sip.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=z9hG4bK5f6de3da5aeaa;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 106 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307113 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:09:12.517] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:48 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 226 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6df2c0f6105 CSeq: 106 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 11 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 <-------------> [2009-07-02 16:09:12.517] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:09:12.518] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.10:4000 [2009-07-02 16:09:12.518] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:09:12.518] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> INVITE sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:52 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Supported: timer,replaces Min-SE: 1800 Remote-Party-ID: "bbbbbbbbbbb 6906" ;party=calling;screen=yes;privacy=off Cisco-Guid: 890017280-2764164037-1127887-168470720 Content-Length: 0 User-Agent: Cisco-CUCM6.1 To: "6105" ;tag=as74b09d4c Contact: Expires: 180 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6e0d824697 CSeq: 107 INVITE Session-Expires: 1800;refresher=uas Max-Forwards: 70 <-------------> [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.c: --- (19 headers 0 lines) --- [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.c: Sending to 192.168.10.10 : 5060 (no NAT) [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.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=z9hG4bK5f6e0d824697;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 107 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.c: Audio is at 192.168.130.140 port 17366 [2009-07-02 16:09:15.974] VERBOSE[10140] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-07-02 16:09:15.975] VERBOSE[10140] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-07-02 16:09:15.975] VERBOSE[10140] chan_sip.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=z9hG4bK5f6e0d824697;received=192.168.10.10 From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 To: "6105" ;tag=as74b09d4c Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 107 INVITE Server: Asterisk PBX 1.6.1.0 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: 240 v=0 o=root 555307107 555307114 IN IP4 192.168.130.140 s=Asterisk PBX 1.6.1.0 c=IN IP4 192.168.130.140 t=0 0 m=audio 17366 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 <------------> [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> ACK sip:6105@192.168.130.140:5060 SIP/2.0 Date: Thu, 02 Jul 2009 15:01:52 GMT From: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Allow-Events: presence, kpml Content-Length: 226 To: "6105" ;tag=as74b09d4c Content-Type: application/sdp Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5f6e1b1b753e CSeq: 107 ACK Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 13 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 <-------------> [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: --- (11 headers 11 lines) --- [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Found RTP audio format 0 [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Found RTP audio format 101 [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Found audio description format PCMU for ID 0 [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-07-02 16:09:15.980] VERBOSE[10140] chan_sip.c: Peer audio RTP is at port 192.168.10.53:4000 [2009-07-02 16:09:15.980] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:09:15.980] VERBOSE[10893] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6105-09d6b5e8 [2009-07-02 16:10:07.045] NOTICE[10140] chan_sip.c: Received SIP subscribe for peer without mailbox: djw-messenger [2009-07-02 16:10:08.606] VERBOSE[10140] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: OPTIONS sip:192.168.10.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5cf79ef3;rport Max-Forwards: 70 From: "asterisk" ;tag=as224eaac7 To: Contact: Call-ID: 15f8bc5b163a649264a5b4591ea22b1e@192.168.130.140 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 02 Jul 2009 15:10:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [2009-07-02 16:10:08.638] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:02:44 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS From: "asterisk" ;tag=as224eaac7 Content-Length: 0 To: ;tag=1973151278 Call-ID: 15f8bc5b163a649264a5b4591ea22b1e@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK5cf79ef3;rport CSeq: 102 OPTIONS <-------------> [2009-07-02 16:10:08.638] VERBOSE[10140] chan_sip.c: --- (9 headers 0 lines) --- [2009-07-02 16:10:08.639] VERBOSE[10140] chan_sip.c: Really destroying SIP dialog '15f8bc5b163a649264a5b4591ea22b1e@192.168.130.140' Method: OPTIONS [2009-07-02 16:10:08.681] VERBOSE[10140] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.10:5060: OPTIONS sip:192.168.10.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK71f6db93;rport Max-Forwards: 70 From: "asterisk" ;tag=as6617d553 To: Contact: Call-ID: 04c0ceb35b2f12ff03e952032cc6f246@192.168.130.140 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 02 Jul 2009 15:10:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [2009-07-02 16:10:08.682] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:02:44 GMT Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS From: "asterisk" ;tag=as6617d553 Content-Length: 0 To: ;tag=1245013004 Call-ID: 04c0ceb35b2f12ff03e952032cc6f246@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK71f6db93;rport CSeq: 102 OPTIONS <-------------> [2009-07-02 16:10:08.682] VERBOSE[10140] chan_sip.c: --- (9 headers 0 lines) --- [2009-07-02 16:10:08.682] VERBOSE[10140] chan_sip.c: Really destroying SIP dialog '04c0ceb35b2f12ff03e952032cc6f246@192.168.130.140' Method: OPTIONS [2009-07-02 16:10:16.823] NOTICE[10140] chan_sip.c: Disconnecting call 'SIP/cisco1-09d71568' for lack of RTP activity in 61 seconds [2009-07-02 16:10:16.830] VERBOSE[10893] chan_sip.c: Scheduling destruction of SIP dialog '750d23b641550a3d7b853c8a058f4500@192.168.130.140' in 6400 ms (Method: ACK) [2009-07-02 16:10:16.831] VERBOSE[10893] chan_sip.c: set_destination: Parsing for address/port to send to [2009-07-02 16:10:16.832] VERBOSE[10893] chan_sip.c: set_destination: set destination to 192.168.10.10, port 5060 [2009-07-02 16:10:16.832] VERBOSE[10893] chan_sip.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=z9hG4bK2e14e30f;rport Max-Forwards: 70 From: "6105" ;tag=as74b09d4c To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2009-07-02 16:10:16.833] VERBOSE[10893] pbx.c: == Spawn extension (aaaaaaaaaaaaaaaa, 6906, 52005) exited non-zero on 'SIP/6105-09d6b5e8' [2009-07-02 16:10:16.834] VERBOSE[10893] res_musiconhold.c: -- Stopped music on hold on SIP/6105-09d6b5e8 [2009-07-02 16:10:16.838] VERBOSE[10140] chan_sip.c: <--- SIP read from UDP://192.168.10.10:5060 ---> SIP/2.0 200 OK Date: Thu, 02 Jul 2009 15:02:53 GMT From: "6105" ;tag=as74b09d4c Content-Length: 0 To: ;tag=dafadd12-25b2-40d3-958c-bc3d033e7ed0-20721585 Call-ID: 750d23b641550a3d7b853c8a058f4500@192.168.130.140 Via: SIP/2.0/UDP 192.168.130.140:5060;branch=z9hG4bK2e14e30f;rport CSeq: 103 BYE