<-------------> --- (11 headers 0 lines) --- [Ktampa3*CLI> <--- SIP read from 74.100.136.230:1311 ---> INVITE sip:4166289921@tampa3.mixnetworks.net:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: CSeq: 2 INVITE Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="50ca84fc0a8b738de10f12e71b736c5e", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1165029551 1165029551 IN IP4 192.168.1.103 s=Polycom IP Phone c=IN IP4 192.168.1.103 t=0 0 a=sendrecv m=audio 2264 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 <-------------> --- (15 headers 11 lines) --- Sending to 74.100.136.230 : 1311 (NAT) Using INVITE request as basis request - fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 Found user 'lmadsen5#lmentinc' [Ktampa3*CLI> Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.103:2264 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.103:2264 Looking for 4166289921 in origination (domain tampa3.mixnetworks.net) list_route: hop: <--- Transmitting (NAT) to 74.100.136.230:1311 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230 From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Ktampa3*CLI> Restrict CID: YES [Ktampa3*CLI> Really destroying SIP dialog '4ba566434a4c6da9654b168479c9676e@8.7.177.3' Method: INVITE [Ktampa3*CLI> Scheduling destruction of SIP dialog '072fec8d5dc80c1b1b49cc1f15f412c4@8.7.177.3' in 6400 ms (Method: INVITE) Audio is at 8.7.177.3 port 12056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 8.7.177.102:5060: INVITE sip:38811278614166289921@8.7.177.102 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK26e1f821;rport From: "LM Enterprises" ;tag=as7b869eba To: Contact: Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "LM Enterprises" ;privacy=off;screen=no Date: Sat, 02 Dec 2006 03:19:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 305 v=0 o=root 21383 21383 IN IP4 8.7.177.3 s=session c=IN IP4 8.7.177.3 t=0 0 m=audio 12056 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Ktampa3*CLI> <--- SIP read from 8.7.177.102:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 8.7.177.3;branch=z9hG4bK26e1f821;received=8.7.177.3;rport=5060 From: "LM Enterprises" ;tag=as7b869eba To: ;tag=cba-03fa-4570b6af Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3 CSeq: 102 INVITE Date: Fri, 01 Dec 2006 23:11:43 GMT Server: BRSIP v2.0.0.11 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Ktampa3*CLI> <--- SIP read from 8.7.177.101:5060 ---> INVITE sip:14166289921@8.7.177.3 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051 From: ;tag=cba-a8b6-4570f051 To: Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101 CSeq: 1165029458 INVITE Contact: Date: Sat, 02 Dec 2006 03:17:37 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 900 Expires: 300 Content-Type: application/sdp Content-Length: 274 v=0 o=BRSDP 501942 501942 IN IP4 66.59.182.210 s=BRSDP Session c=IN IP4 66.59.182.210 t=0 0 m=audio 17168 RTP/AVP 0 18 4 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (18 headers 12 lines) --- Sending to 8.7.177.101 : 5060 (no NAT) Using INVITE request as basis request - 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101 Found peer 'telic101' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 66.59.182.210:17168 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.59.182.210:17168 Looking for 14166289921 in from-voip (domain 8.7.177.3) list_route: hop: <--- Transmitting (no NAT) to 8.7.177.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051;received=8.7.177.101 From: ;tag=cba-a8b6-4570f051 To: Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101 CSeq: 1165029458 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Ktampa3*CLI> Scheduling destruction of SIP dialog '35da68b8301b518962f0e4786965fa54@8.7.177.3' in 13056 ms (Method: INVITE) [Ktampa3*CLI> Audio is at 8.7.177.3 port 16182 [Ktampa3*CLI> Adding codec 0x100 (g729) to SDP [Ktampa3*CLI> Adding codec 0x2 (gsm) to SDP [Ktampa3*CLI> Adding codec 0x4 (ulaw) to SDP [Ktampa3*CLI> Adding codec 0x8 (alaw) to SDP [Ktampa3*CLI> Adding codec 0x10 (g726aal2) to SDP [Ktampa3*CLI> Adding codec 0x20 (adpcm) to SDP [Ktampa3*CLI> Adding codec 0x40 (slin) to SDP [Ktampa3*CLI> Adding codec 0x80 (lpc10) to SDP [Ktampa3*CLI> Adding codec 0x400 (ilbc) to SDP [Ktampa3*CLI> Adding codec 0x800 (g726) to SDP [Ktampa3*CLI> Adding non-codec 0x1 (telephone-event) to SDP [Ktampa3*CLI> Reliably Transmitting (NAT) to 74.100.136.230:1212: INVITE sip:lmadsen%23lmentinc@192.168.1.100:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport From: "4166289921" ;tag=as3f6ac536 To: Contact: Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "4166289921" ;privacy=off;screen=no Date: Sat, 02 Dec 2006 03:19:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 512 v=0 o=root 21383 21383 IN IP4 8.7.177.3 s=session c=IN IP4 8.7.177.3 t=0 0 m=audio 16182 RTP/AVP 18 3 0 8 112 5 10 7 97 111 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Ktampa3*CLI> <--- SIP read from 74.100.136.230:1212 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport From: "4166289921" ;tag=as3f6ac536 To: Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3 Date: Sat, 02 Dec 2006 03:19:20 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Ktampa3*CLI> <--- SIP read from 74.100.136.230:1212 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK48dc73dd;rport From: "4166289921" ;tag=as3f6ac536 To: ;tag=00036bdd39202a341e2b724f-43ee9bc1 Call-ID: 7d95b4ca7c8bc44451b038d32b691538@8.7.177.3 Date: Sat, 02 Dec 2006 03:19:20 GMT CSeq: 102 INVITE Server: Cisco-CP7960G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Ktampa3*CLI> <--- Transmitting (no NAT) to 8.7.177.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 8.7.177.101;branch=z9hG4bKf6f6d3ff6ff051;received=8.7.177.101 From: ;tag=cba-a8b6-4570f051 To: ;tag=as4fc2ecf2 Call-ID: 8f751680c65cd745-a8b6-4570f051-86e7@8.7.177.101 CSeq: 1165029458 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Ktampa3*CLI> <--- SIP read from 8.7.177.102:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 8.7.177.3;branch=z9hG4bK26e1f821;received=8.7.177.3;rport=5060 From: "LM Enterprises" ;tag=as7b869eba To: ;tag=cba-03fa-4570b6af Call-ID: 548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3 CSeq: 102 INVITE Contact: Date: Fri, 01 Dec 2006 23:11:47 GMT Server: BRSIP v2.0.0.11 Content-Type: application/sdp Content-Length: 198 v=0 o=BRSDP 1018 1018 IN IP4 209.244.42.249 s=BRSDP Session c=IN IP4 209.244.42.249 t=0 0 m=audio 62984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.244.42.249:62984 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.249:62984 [Ktampa3*CLI> Audio is at 8.7.177.3 port 17040 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 74.100.136.230:1311 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230 From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: ;tag=as7a40e8d1 Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 21383 21383 IN IP4 8.7.177.3 s=session c=IN IP4 8.7.177.3 t=0 0 m=audio 17040 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Ktampa3*CLI> Reliably Transmitting (no NAT) to 69.17.122.217:5060: OPTIONS sip:69.17.122.217 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport From: "asterisk" ;tag=as23c24796 To: Contact: Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces [Ktampa3*CLI> Content-Length: 0 --- [Ktampa3*CLI> Retransmitting #1 (no NAT) to 69.17.122.217:5060: OPTIONS sip:69.17.122.217 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport From: "asterisk" ;tag=as23c24796 To: Contact: Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Ktampa3*CLI> Really destroying SIP dialog '072fec8d5dc80c1b1b49cc1f15f412c4@8.7.177.3' Method: INVITE [Ktampa3*CLI> Retransmitting #2 (no NAT) to 69.17.122.217:5060: OPTIONS sip:69.17.122.217 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport From: "asterisk" ;tag=as23c24796 To: Contact: Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Ktampa3*CLI> Retransmitting #3 (no NAT) to 69.17.122.217:5060: OPTIONS sip:69.17.122.217 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK1116f9c2;rport From: "asterisk" ;tag=as23c24796 To: Contact: Call-ID: 36a9d1bd16da06561260a5110148d96b@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Ktampa3*CLI> <--- SIP read from 74.100.136.230:1311 ---> CANCEL sip:4166289921@tampa3.mixnetworks.net:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: CSeq: 2 CANCEL Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036 Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="50ca84fc0a8b738de10f12e71b736c5e", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 74.100.136.230 : 1311 (NAT) <--- Reliably Transmitting (NAT) to 74.100.136.230:1311 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230 From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: ;tag=as7a40e8d1 Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 74.100.136.230:1311 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A;received=74.100.136.230 From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: ;tag=as7a40e8d1 Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Ktampa3*CLI> Scheduling destruction of SIP dialog '548183ff45f2ff4a3ec7edac6890b98e@8.7.177.3' in 6400 ms (Method: INVITE) [Ktampa3*CLI> <--- SIP read from 74.100.136.230:1311 ---> ACK sip:4166289921@8.7.177.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK3e54eef3C401F11A From: "lmadsen5#lmentinc" ;tag=1D2CE228-54CF5249 To: ;tag=as7a40e8d1 CSeq: 2 ACK Call-ID: fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036 Proxy-Authorization: Digest username="lmadsen5#lmentinc", realm="asterisk", nonce="440b8862", uri="sip:4166289921@tampa3.mixnetworks.net:5060;user=phone", response="afca69fef6bfde509fbd878c65218fda", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog 'fd3e7c14-89fdf4a6-a12dc54f@192.168.1.103' Method: ACK [Ktampa3*CLI> Reliably Transmitting (no NAT) to 8.7.177.99:5060: OPTIONS sip:8.7.177.99 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK60ce581c;rport From: "asterisk" ;tag=as4f756162 To: Contact: Call-ID: 36d3561d7cc655673b8e4313241efca1@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Ktampa3*CLI> <--- SIP read from 8.7.177.99:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK60ce581c;rport=5060 From: "asterisk" ;tag=as4f756162 To: ;tag=f838092f89d25b94c908dbbca208fb0a.075a Call-ID: 36d3561d7cc655673b8e4313241efca1@8.7.177.3 CSeq: 102 OPTIONS Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 8.7.177.99:5060 "Noisy feedback tells: pid=3775 req_src_ip=8.7.177.3 req_src_port=5060 in_uri=sip:8.7.177.99 out_uri=sip:8.7.177.99 via_cnt==1" <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '36d3561d7cc655673b8e4313241efca1@8.7.177.3' Method: OPTIONS [Ktampa3*CLI> Reliably Transmitting (no NAT) to 8.7.177.2:5060: OPTIONS sip:8.7.177.2 SIP/2.0 Via: SIP/2.0/UDP 8.7.177.3:5060;branch=z9hG4bK537ae707;rport From: "asterisk" ;tag=as02815eee To: Contact: Call-ID: 0ceb0d975cadfa897bbb7be30a259a73@8.7.177.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 02 Dec 2006 03:19:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0