sip debug SIP Debugging enabled [Sep 29 18:57:05] 11 headers, 0 lines [Sep 29 18:57:05] Reliably Transmitting (no NAT) to 192.168.1.113:5060: OPTIONS sip:113@192.168.1.113:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK206c224f From: "asterisk" ;tag=as354dff5a To: Contact: Call-ID: 0652c276263b34273672cd6f4412d630@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:05] 11 headers, 0 lines [Sep 29 18:57:05] Reliably Transmitting (no NAT) to 192.168.1.110:5060: OPTIONS sip:110@192.168.1.110:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK377e8d11 From: "asterisk" ;tag=as5f914827 To: Contact: Call-ID: 1cc9c37023ece8423c7b26f86b98b3f3@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:05] <-- SIP read from 192.168.1.113:51178: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK206c224f From: "asterisk" ;tag=as354dff5a To: ;tag=001201dbc2835ad136ebcc1b-69a50464 Call-ID: 0652c276263b34273672cd6f4412d630@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:08 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 216 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 1844 0 IN IP4 192.168.1.113 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:05] --- (18 headers 10 lines)[Sep 29 18:57:05] --- [Sep 29 18:57:05] Destroying call '0652c276263b34273672cd6f4412d630@192.168.1.14' [Sep 29 18:57:06] <-- SIP read from 192.168.1.110:50619: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK377e8d11 From: "asterisk" ;tag=as5f914827 To: ;tag=001201dbe06c00de3804f602-50082da3 Call-ID: 1cc9c37023ece8423c7b26f86b98b3f3@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:09 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 215 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 916 0 IN IP4 192.168.1.110 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:06] --- (18 headers 10 lines)[Sep 29 18:57:06] --- [Sep 29 18:57:06] Destroying call '1cc9c37023ece8423c7b26f86b98b3f3@192.168.1.14' [Sep 29 18:57:06] 11 headers, 0 lines [Sep 29 18:57:06] Reliably Transmitting (no NAT) to 192.168.1.102:5060: OPTIONS sip:102@192.168.1.102:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK12a47ea6 From: "asterisk" ;tag=as730c8bf0 To: Contact: Call-ID: 17de22170a5e869242dd0db7184e057d@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:06] <-- SIP read from 192.168.1.102:52617: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK12a47ea6 From: "asterisk" ;tag=as730c8bf0 To: ;tag=001201dbe20c596d2994feb1-73a4b038 Call-ID: 17de22170a5e869242dd0db7184e057d@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:09 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 217 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 24923 0 IN IP4 192.168.1.102 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:06] --- (18 headers 10 lines)[Sep 29 18:57:06] --- [Sep 29 18:57:06] Destroying call '17de22170a5e869242dd0db7184e057d@192.168.1.14' [Sep 29 18:57:06] 11 headers, 0 lines [Sep 29 18:57:06] Reliably Transmitting (no NAT) to 192.168.1.100:5060: OPTIONS sip:100@192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK49ff831f From: "asterisk" ;tag=as6fdaea9a To: Contact: Call-ID: 7376dda53a40834031f1639744e37c26@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:06] <-- SIP read from 192.168.1.100:50433: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK49ff831f From: "asterisk" ;tag=as6fdaea9a To: ;tag=001200a769b700f35f880980-764fe33a Call-ID: 7376dda53a40834031f1639744e37c26@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:09 GMT CSeq: 102 OPTIONS Server: Cisco-CP7960G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 216 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 1127 0 IN IP4 192.168.1.100 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:06] --- (18 headers 10 lines)[Sep 29 18:57:06] --- [Sep 29 18:57:06] Destroying call '7376dda53a40834031f1639744e37c26@192.168.1.14' [Sep 29 18:57:10] <-- SIP read from 192.168.1.97:6882: [Sep 29 18:57:10] --- (0 headers 0 lines)[Sep 29 18:57:10] Nat keepalive [Sep 29 18:57:10] --- [Sep 29 18:57:14] <-- SIP read from 192.168.1.97:6882: INVITE sip:105@asterisk02.katherinebishop.com SIP/2.0 To: From: "Serge (eyebeam)";tag=232bb74e Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-965765868-1--d87543-;rport Call-ID: aa6382040e54d91e CSeq: 1 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3008q stamp 18440 Content-Length: 325 v=0 o=- 108505955 108505977 IN IP4 192.168.1.97 s=eyeBeam c=IN IP4 192.168.1.97 t=0 0 m=audio 10574 RTP/AVP 100 6 0 8 3 98 97 5 101 a=alt:1 1 : 9D02E1F1 0000007B 192.168.1.97 10574 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:98 ilbc/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv [Sep 29 18:57:14] --- (12 headers 13 lines)[Sep 29 18:57:14] --- [Sep 29 18:57:14] Using INVITE request as basis request - aa6382040e54d91e [Sep 29 18:57:14] Sending to 192.168.1.97 : 6882 (non-NAT) [Sep 29 18:57:14] Reliably Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-965765868-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: ;tag=as3c0b7b4b Call-ID: aa6382040e54d91e CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk02.katherinebishop.com", nonce="05d2c32c" Content-Length: 0 --- [Sep 29 18:57:14] Scheduling destruction of call 'aa6382040e54d91e' in 15000 ms [Sep 29 18:57:14] Found user '151' [Sep 29 18:57:14] <-- SIP read from 192.168.1.97:6882: ACK sip:105@asterisk02.katherinebishop.com SIP/2.0 To: ;tag=as3c0b7b4b From: "Serge (eyebeam)";tag=232bb74e Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-965765868-1--d87543-;rport Call-ID: aa6382040e54d91e CSeq: 1 ACK Content-Length: 0 [Sep 29 18:57:14] --- (7 headers 0 lines)[Sep 29 18:57:14] --- [Sep 29 18:57:14] <-- SIP read from 192.168.1.97:6882: INVITE sip:105@asterisk02.katherinebishop.com SIP/2.0 To: From: "Serge (eyebeam)";tag=232bb74e Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport Call-ID: aa6382040e54d91e CSeq: 2 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="151",realm="asterisk02.katherinebishop.com",nonce="05d2c32c",uri="sip:105@asterisk02.katherinebishop.com",response="31c80c0ac105ae0b4a66cb977e11232f",algorithm=MD5 User-Agent: eyeBeam release 3008q stamp 18440 Content-Length: 325 v=0 o=- 108505955 108505977 IN IP4 192.168.1.97 s=eyeBeam c=IN IP4 192.168.1.97 t=0 0 m=audio 10574 RTP/AVP 100 6 0 8 3 98 97 5 101 a=alt:1 1 : 9D02E1F1 0000007B 192.168.1.97 10574 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:98 ilbc/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv [Sep 29 18:57:14] --- (13 headers 13 lines)[Sep 29 18:57:14] --- [Sep 29 18:57:14] Using INVITE request as basis request - aa6382040e54d91e [Sep 29 18:57:14] Sending to 192.168.1.97 : 6882 (non-NAT) [Sep 29 18:57:14] Found user '151' [Sep 29 18:57:14] Found RTP audio format 100 [Sep 29 18:57:14] Found RTP audio format 6 [Sep 29 18:57:14] Found RTP audio format 0 [Sep 29 18:57:14] Found RTP audio format 8 [Sep 29 18:57:14] Found RTP audio format 3 [Sep 29 18:57:14] Found RTP audio format 98 [Sep 29 18:57:14] Found RTP audio format 97 [Sep 29 18:57:14] Found RTP audio format 5 [Sep 29 18:57:14] Found RTP audio format 101 [Sep 29 18:57:14] Peer audio RTP is at port 192.168.1.97:10574 [Sep 29 18:57:14] Peer video RTP is at port 192.168.1.97:65535 [Sep 29 18:57:14] Found description format speex [Sep 29 18:57:14] Found description format ilbc [Sep 29 18:57:14] Found description format speex [Sep 29 18:57:14] Found description format telephone-event [Sep 29 18:57:14] Capabilities: us - 0x80406 (gsm|ulaw|ilbc|h263), peer - audio=0x62e (gsm|ulaw|alaw|adpcm|speex|ilbc)/video=0x0 (nothing), combined - 0x406 (gsm|ulaw|ilbc) [Sep 29 18:57:14] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Sep 29 18:57:14] Looking for 105 in ldaccess (domain asterisk02.katherinebishop.com) [Sep 29 18:57:14] list_route: hop: [Sep 29 18:57:14] Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: Call-ID: aa6382040e54d91e CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [Sep 29 18:57:14] -- Executing Macro("SIP/151-7e9f", "stdexten|105|SIP/105") in new stack [Sep 29 18:57:14] -- Executing NoOp("SIP/151-7e9f", "extension s@asterisk02.katherinebishop.com") in new stack [Sep 29 18:57:14] -- Executing Dial("SIP/151-7e9f", "SIP/105|20") in new stack [Sep 29 18:57:14] We're at 192.168.1.14 port 18980 [Sep 29 18:57:14] Video is at 192.168.1.14 port 11728 [Sep 29 18:57:14] Adding codec 0x2 (gsm) to SDP [Sep 29 18:57:14] Adding codec 0x4 (ulaw) to SDP [Sep 29 18:57:14] Adding codec 0x400 (ilbc) to SDP [Sep 29 18:57:14] Adding codec 0x80000 (h263) to SDP [Sep 29 18:57:14] Adding non-codec 0x1 (telephone-event) to SDP [Sep 29 18:57:14] 12 headers, 14 lines [Sep 29 18:57:14] Reliably Transmitting (no NAT) to 192.168.1.105:5060: INVITE sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: Contact: Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 313 v=0 o=root 7730 7730 IN IP4 192.168.1.14 s=session c=IN IP4 192.168.1.14 t=0 0 m=audio 18980 RTP/AVP 3 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 11728 RTP/AVP 34 a=rtpmap:34 H263/90000 --- [Sep 29 18:57:14] -- Called 105 [Sep 29 18:57:15] <-- SIP read from 192.168.1.105:50615: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:18 GMT CSeq: 102 INVITE Server: Cisco-CP7940G/7.5 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 [Sep 29 18:57:15] --- (11 headers 0 lines)[Sep 29 18:57:15] --- [Sep 29 18:57:15] <-- SIP read from 192.168.1.105:50615: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: ;tag=001201dbe43d0f3d4e55c41b-480d61a7 Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:18 GMT CSeq: 102 INVITE Server: Cisco-CP7940G/7.5 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 [Sep 29 18:57:15] --- (11 headers 0 lines)[Sep 29 18:57:15] --- [Sep 29 18:57:15] -- SIP/105-3e2c is ringing [Sep 29 18:57:15] Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: ;tag=as7724ab44 Call-ID: aa6382040e54d91e CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [Sep 29 18:57:15] We're at 192.168.1.14 port 10236 [Sep 29 18:57:15] Video is at 192.168.1.14 port 16238 [Sep 29 18:57:15] Adding codec 0x2 (gsm) to SDP [Sep 29 18:57:15] Adding codec 0x4 (ulaw) to SDP [Sep 29 18:57:15] Adding codec 0x400 (ilbc) to SDP [Sep 29 18:57:15] Adding codec 0x80000 (h263) to SDP [Sep 29 18:57:15] Adding non-codec 0x1 (telephone-event) to SDP [Sep 29 18:57:15] Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: ;tag=as7724ab44 Call-ID: aa6382040e54d91e CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 7730 7730 IN IP4 192.168.1.14 s=session c=IN IP4 192.168.1.14 t=0 0 m=audio 10236 RTP/AVP 3 0 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Sep 29 18:57:16] <-- SIP read from 192.168.1.97:6882: CANCEL sip:105@asterisk02.katherinebishop.com SIP/2.0 To: From: "Serge (eyebeam)";tag=232bb74e Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport Call-ID: aa6382040e54d91e CSeq: 2 CANCEL Proxy-Authorization: Digest username="151",realm="asterisk02.katherinebishop.com",nonce="05d2c32c",uri="sip:105@asterisk02.katherinebishop.com",response="15cb9fae155207c4c15c58f5c8508df9",algorithm=MD5 User-Agent: eyeBeam release 3008q stamp 18440 Content-Length: 0 [Sep 29 18:57:16] --- (9 headers 0 lines)[Sep 29 18:57:16] --- [Sep 29 18:57:16] Sending to 192.168.1.97 : 6882 (non-NAT) [Sep 29 18:57:16] Reliably Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: ;tag=as7724ab44 Call-ID: aa6382040e54d91e CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [Sep 29 18:57:16] Transmitting (no NAT) to 192.168.1.97:6882: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport;received=192.168.1.97 From: "Serge (eyebeam)";tag=232bb74e To: ;tag=as7724ab44 Call-ID: aa6382040e54d91e CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [Sep 29 18:57:16] Reliably Transmitting (no NAT) to 192.168.1.105:5060: CANCEL sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: Contact: Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 --- [Sep 29 18:57:16] Scheduling destruction of call '40952b786b43baec7a377101340c091e@192.168.1.14' in 15000 ms [Sep 29 18:57:16] -- Executing Hangup("SIP/151-7e9f", "") in new stack [Sep 29 18:57:16] <-- SIP read from 192.168.1.97:6882: ACK sip:105@asterisk02.katherinebishop.com SIP/2.0 To: ;tag=as7724ab44 From: "Serge (eyebeam)";tag=232bb74e Via: SIP/2.0/UDP 192.168.1.97:6882;branch=z9hG4bK-d87543-492440791-1--d87543-;rport Call-ID: aa6382040e54d91e CSeq: 2 ACK Content-Length: 0 [Sep 29 18:57:16] --- (7 headers 0 lines)[Sep 29 18:57:16] --- [Sep 29 18:57:16] Destroying call 'aa6382040e54d91e' [Sep 29 18:57:16] <-- SIP read from 192.168.1.105:50615: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: ;tag=001201dbe43d0f3d4e55c41b-480d61a7 Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:19 GMT CSeq: 102 CANCEL Server: Cisco-CP7940G/7.5 Content-Length: 0 [Sep 29 18:57:16] --- (9 headers 0 lines)[Sep 29 18:57:16] --- [Sep 29 18:57:16] <-- SIP read from 192.168.1.105:50615: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: ;tag=001201dbe43d0f3d4e55c41b-480d61a7 Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:19 GMT CSeq: 102 INVITE Server: Cisco-CP7940G/7.5 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 [Sep 29 18:57:16] --- (11 headers 0 lines)[Sep 29 18:57:16] --- [Sep 29 18:57:16] Transmitting (no NAT) to 192.168.1.105:5060: ACK sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK47393b7a;rport From: "Serge Vecher (videophone)" ;tag=as550efbac To: ;tag=001201dbe43d0f3d4e55c41b-480d61a7 Contact: Call-ID: 40952b786b43baec7a377101340c091e@192.168.1.14 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- [Sep 29 18:57:16] Destroying call '40952b786b43baec7a377101340c091e@192.168.1.14' [Sep 29 18:57:17] 11 headers, 0 lines [Sep 29 18:57:17] Reliably Transmitting (no NAT) to 192.168.1.105:5060: OPTIONS sip:105@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK1852c6f0 From: "asterisk" ;tag=as3f560994 To: Contact: Call-ID: 0dd03f4b5aea80877d8b1e53523e31fa@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:17] <-- SIP read from 192.168.1.105:51146: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK1852c6f0 From: "asterisk" ;tag=as3f560994 To: ;tag=001201dbe43d0f3e5b28bb8c-5507c6d2 Call-ID: 0dd03f4b5aea80877d8b1e53523e31fa@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:21 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 217 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 13597 0 IN IP4 192.168.1.105 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:17] --- (18 headers 10 lines)[Sep 29 18:57:17] --- [Sep 29 18:57:17] Destroying call '0dd03f4b5aea80877d8b1e53523e31fa@192.168.1.14' [Sep 29 18:57:17] 11 headers, 0 lines [Sep 29 18:57:17] Reliably Transmitting (no NAT) to 192.168.1.105:5060: OPTIONS sip:serge@192.168.1.105:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK0168f68d From: "asterisk" ;tag=as09b8b09a To: Contact: Call-ID: 36abe308792fdecc6fb7cb723be70649@192.168.1.14 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 29 Sep 2005 22:57:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- [Sep 29 18:57:17] <-- SIP read from 192.168.1.105:51147: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK0168f68d From: "asterisk" ;tag=as09b8b09a To: ;tag=001201dbe43d0f3f600984d6-22812a3b Call-ID: 36abe308792fdecc6fb7cb723be70649@192.168.1.14 Date: Thu, 29 Sep 2005 22:57:21 GMT CSeq: 102 OPTIONS Server: Cisco-CP7940G/7.5 Content-Length: 0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces Content-Length: 217 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 11657 0 IN IP4 192.168.1.105 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Sep 29 18:57:17] --- (18 headers 10 lines)[Sep 29 18:57:17] --- [Sep 29 18:57:17] Destroying call '36abe308792fdecc6fb7cb723be70649@192.168.1.14' [Sep 29 18:57:19] <-- SIP read from 192.168.1.97:6882: [Sep 29 18:57:19] --- (0 headers 0 lines)[Sep 29 18:57:19] Nat keepalive [Sep 29 18:57:19] --- sip no debug SIP Debugging Disabled *CLI>