pbx*CLI> sip debug peer testphone SIP Debugging Enabled for IP: 72.187.95.191:58099 pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> INVITE sip:15552937963@pbx.elephantoutlook.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK2b9695ccDEAD35E3 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: CSeq: 1 INVITE Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1190836785 1190836785 IN IP4 192.168.2.21 s=Polycom IP Phone c=IN IP4 192.168.2.21 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Using INVITE request as basis request - 1a5f46d2-f4de6c59-a267b038@192.168.2.21 <--- Reliably Transmitting (NAT) to 72.187.95.191:58099 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK2b9695ccDEAD35E3;received=72.187.95.191 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as28a3e91c Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4605e8a7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1a5f46d2-f4de6c59-a267b038@192.168.2.21' in 32000 ms (Method: INVITE) Found user 'testphone' pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> ACK sip:15552937963@pbx.elephantoutlook.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK2b9695ccDEAD35E3 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as28a3e91c CSeq: 1 ACK Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> INVITE sip:15552937963@pbx.elephantoutlook.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK3fd53d651FD82424 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: CSeq: 2 INVITE Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="testphone", realm="asterisk", nonce="4605e8a7", uri="sip:15552937963@pbx.elephantoutlook.com:5060;user=phone", response="450a3fb801c0903c68f7b12cccc8e5a2", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1190836785 1190836785 IN IP4 192.168.2.21 s=Polycom IP Phone c=IN IP4 192.168.2.21 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 a=sendrecv 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 72.187.95.191 : 58099 (NAT) Using INVITE request as basis request - 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Found user 'testphone' 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 192.168.2.21:2222 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 - 0x80006 (gsm|ulaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/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 192.168.2.21:2222 Looking for 15552937963 in local (domain pbx.elephantoutlook.com) list_route: hop: pbx*CLI> <--- Transmitting (NAT) to 72.187.95.191:58099 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK3fd53d651FD82424;received=72.187.95.191 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [dial-15552937963@local:1] Dial("SIP/testphone-084fecf8", "SIP/bandwidth/+15552937963") in new stack -- Called bandwidth/+15552937963 [Sep 26 16:23:16] WARNING[25857]: chan_sip.c:5017 process_sdp: Unsupported SDP media type in offer: image 34618 udptl t38 -- SIP/bandwidth-0844ae70 is making progress passing it to SIP/testphone-084fecf8 Audio is at 192.168.10.225 port 16654 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 72.187.95.191:58099 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK3fd53d651FD82424;received=72.187.95.191 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as06093069 Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 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: 244 v=0 o=root 16400 16400 IN IP4 192.168.10.225 s=session c=IN IP4 192.168.10.225 t=0 0 m=audio 16654 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Really destroying SIP dialog '4d76075349950d680009c5a6695bf6ca@192.168.10.225' Method: OPTIONS [Sep 26 16:23:19] WARNING[25857]: chan_sip.c:5017 process_sdp: Unsupported SDP media type in offer: image 34618 udptl t38 -- SIP/bandwidth-0844ae70 answered SIP/testphone-084fecf8 Audio is at 192.168.10.225 port 16654 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK3fd53d651FD82424;received=72.187.95.191 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as06093069 Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 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: 244 v=0 o=root 16400 16401 IN IP4 192.168.10.225 s=session c=IN IP4 192.168.10.225 t=0 0 m=audio 16654 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> ACK sip:15552937963@192.168.10.225 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bKf000a1907BB8FB47 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as06093069 CSeq: 2 ACK Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Proxy-Authorization: Digest username="testphone", realm="asterisk", nonce="4605e8a7", uri="sip:15552937963@pbx.elephantoutlook.com:5060;user=phone", response="450a3fb801c0903c68f7b12cccc8e5a2", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '4adae41c1e3c087033f770b70fd39fe5@192.168.10.225' Method: OPTIONS pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> BYE sip:15552937963@192.168.10.225 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK99437ed6D94D77FD From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as06093069 CSeq: 3 BYE Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 Contact: User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Proxy-Authorization: Digest username="testphone", realm="asterisk", nonce="4605e8a7", uri="sip:15552937963@pbx.elephantoutlook.com:5060;user=phone", response="40f0b386776303daf00a5c507a465876", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 72.187.95.191 : 58099 (NAT) <--- Transmitting (NAT) to 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.21;branch=z9hG4bK99437ed6D94D77FD;received=72.187.95.191 From: "Jon Webster" ;tag=E12FF22F-8555D7FE To: ;tag=as06093069 Call-ID: 1a5f46d2-f4de6c59-a267b038@192.168.2.21 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (local, dial-15552937963, 6) exited non-zero on 'SIP/testphone-084fecf8' Really destroying SIP dialog '1a5f46d2-f4de6c59-a267b038@192.168.2.21' Method: BYE [Sep 26 16:23:32] WARNING[25857]: chan_sip.c:12528 handle_response: Remote host can't match request BYE to call '501cbe474ae7bd131da738b751a8c8e8@4.68.250.148'. Giving up. Really destroying SIP dialog '501cbe474ae7bd131da738b751a8c8e8@4.68.250.148' Method: INVITE Really destroying SIP dialog '0c1a137125d36eeb03a926c2307ee5ab@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5b5f5f87;rport From: "asterisk" ;tag=as3159fcb9 To: Contact: Call-ID: 59fcda6645d6f70415f3b54c60db725f@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:23:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5b5f5f87;rport From: "asterisk" ;tag=as3159fcb9 To: ;tag=EFE5A813-7AA7F482 CSeq: 102 OPTIONS Call-ID: 59fcda6645d6f70415f3b54c60db725f@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '59fcda6645d6f70415f3b54c60db725f@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '78f506b90e5690341e9e8d5e59a62c47@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2f37b2011f96f9f13b4170eb4475cdd3@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '102159865afc3da3793846132d2fb6ca@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '79af942653ac6e8c198365902f771adc@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '19848efb127995b420d025e93f6eb347@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '31f4dfe36d97209d1da78f651d77994a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '07352f45175688cc7252c66f6b8b0c82@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '325416871b3e7fd90a39cc32054f2bfc@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK0af94dc1;rport From: "asterisk" ;tag=as1fdacb4b To: Contact: Call-ID: 4aa0de15595155a65300e9cc26ad5858@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:24:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK0af94dc1;rport From: "asterisk" ;tag=as1fdacb4b To: ;tag=2AFA78E8-AB0B5A5F CSeq: 102 OPTIONS Call-ID: 4aa0de15595155a65300e9cc26ad5858@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4aa0de15595155a65300e9cc26ad5858@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '04d53bcc0166264c1aeab5472d644afc@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '36ec71d354640b1d7b320a0a5137baf9@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '0b8df00824d8c0c82c0df00d436478d9@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '017be5814c1cc396256e193d786f0309@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '359b26423a70620f3bd6885b56c13da3@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '608194086498722c5e51541a09ff15eb@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '72d962be7e6752c62340138e4f4e039c@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '16fd15d1063f51432489bdb7752840b2@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK3355abee;rport From: "asterisk" ;tag=as52820d8e To: Contact: Call-ID: 7aa3cf3e7142f70d72bedc8e3edba025@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:25:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK3355abee;rport From: "asterisk" ;tag=as52820d8e To: ;tag=F18C6895-D45B32D4 CSeq: 102 OPTIONS Call-ID: 7aa3cf3e7142f70d72bedc8e3edba025@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '7aa3cf3e7142f70d72bedc8e3edba025@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2254b2a4658ad9fc1b26003e5596624d@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '421ddab74c1cf0a36f60e8fc49915329@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '5333778966a5f2a6597f556457e6a82a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '148a41390f2e107700710785732fe413@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '43e852ee663d05924bc7df8e66eddfcd@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '618188db650227f903b4a86102e8bcd2@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2e9fcc7f01d34408687936ae41f88c12@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '0aa04cc077d250aa7476cc6208185dea@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5be3e000;rport From: "asterisk" ;tag=as13f6c26e To: Contact: Call-ID: 17addc8e1756a8032be0e93c3b0ef9b4@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:26:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5be3e000;rport From: "asterisk" ;tag=as13f6c26e To: ;tag=6BBA945A-B2DE1D21 CSeq: 102 OPTIONS Call-ID: 17addc8e1756a8032be0e93c3b0ef9b4@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '17addc8e1756a8032be0e93c3b0ef9b4@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '17235bfa04463d49482e90372e6b95ca@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1479034b1761c01d7e17038c26ea21b2@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '7790e80c176f4fb1460f1c3147e26039@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '5ebf5d8a766d3a180dc3e21c39a4cb58@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1cd7063a33fa623438409f7d006f2fb5@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1e94f2d843aa3ad80deae4ec39ad17dc@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '055f5d444b6e79751350d9ae43ac7096@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '0747113d4735fb377ef735dc2b2f3ce2@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2934141a;rport From: "asterisk" ;tag=as6c76ce5c To: Contact: Call-ID: 6a73a6c760e0e0e06ea4c2fc28498e54@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:27:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2934141a;rport From: "asterisk" ;tag=as6c76ce5c To: ;tag=A8A08F77-947A1E86 CSeq: 102 OPTIONS Call-ID: 6a73a6c760e0e0e06ea4c2fc28498e54@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '6a73a6c760e0e0e06ea4c2fc28498e54@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '59d90a710dd36ca546140c2246833bd7@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '027418f93c2130d60692a39c657d19ad@192.168.10.225' Method: OPTIONS pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> Really destroying SIP dialog '714e1a345f3e27c6649d850a300bfe7f@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1480e4a47ef48b6b5fd56c4b4e7a2f47@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1b5258ca752b633b02fecfe04912dc03@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2b92dcf35c71d62850fb0a35536f232f@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '754ce998669b03cc45d92b932a76b09d@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '72eab9981c2db340176920eb19eb9fd3@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5e63316a;rport From: "asterisk" ;tag=as2032b6ce To: Contact: Call-ID: 761980c37ee61c2a136700ce125b8c3a@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:28:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK5e63316a;rport From: "asterisk" ;tag=as2032b6ce To: ;tag=AA17B32C-F0E67243 CSeq: 102 OPTIONS Call-ID: 761980c37ee61c2a136700ce125b8c3a@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '761980c37ee61c2a136700ce125b8c3a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '4d023bc22b52905646a4e9203bd04c5c@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '6d5b5c4a49cd32731ac83ca86e375fd7@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '4e7a235e48d818483e2501e124c005ad@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '364c69032c65e9c72b4c05f13eb306c0@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1db89c1b6abad7de160d68345cb25154@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '06e2bf952f626856712ae7ad5e8643f7@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '6f36e8df3db10c3d0689248544ae2666@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '22426dc15b9be2951f9ce5835688a20a@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2fcedd09;rport From: "asterisk" ;tag=as1892d307 To: Contact: Call-ID: 16b13a362dec26cf64388fd3109e799a@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:29:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2fcedd09;rport From: "asterisk" ;tag=as1892d307 To: Contact: Call-ID: 16b13a362dec26cf64388fd3109e799a@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:29:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK2fcedd09;rport From: "asterisk" ;tag=as1892d307 To: ;tag=32C5EEB9-1066D998 CSeq: 102 OPTIONS Call-ID: 16b13a362dec26cf64388fd3109e799a@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '16b13a362dec26cf64388fd3109e799a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '076530077339f11210f28d2d7bad650b@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '7fad0a762f0f72cc203a5a797ec09e71@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '121318616d8eed1a5cc5d5f91a76e236@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2bbb9e40426cd8767058ff4654918f19@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '2c0f954f3374c55726aeb66937a14396@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '5176b0246b93f095645e3b76640b45ed@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '1ee2679b30f57ffc1e846d167b4a430f@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '4acfd2e82e47520f3e8336ba71018925@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7359cb2a;rport From: "asterisk" ;tag=as31129539 To: Contact: Call-ID: 471c0d1801099d842cc53bc46f32143b@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:30:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7359cb2a;rport From: "asterisk" ;tag=as31129539 To: ;tag=954E175E-40046BC5 CSeq: 102 OPTIONS Call-ID: 471c0d1801099d842cc53bc46f32143b@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '471c0d1801099d842cc53bc46f32143b@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '3cf95b1f144f708c405f05dc73d3cb33@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '02d2092c6e65f9c252c4353836cf7b25@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '44adf6c12f4b3d204e2da4bb7f2324b7@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '01b96acd3eb2c5ec5302367913613c55@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '6ecaf8c935e705e136f0a36663b5df2a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '4b1570c37232efdd238e48e626605212@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '501157fd14bf4ebe440a8bdf1238309a@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '478d27a15e9620c1756e3e6b57162efe@192.168.10.225' Method: OPTIONS Reliably Transmitting (NAT) to 72.187.95.191:58099: OPTIONS sip:testphone@192.168.2.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK73cb6523;rport From: "asterisk" ;tag=as74ebed7f To: Contact: Call-ID: 53216ecd6b19edcd59e4e6970fcbec78@192.168.10.225 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 26 Sep 2007 20:31:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 72.187.95.191:58099 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK73cb6523;rport From: "asterisk" ;tag=as74ebed7f To: ;tag=A2D5B85B-E097E60A CSeq: 102 OPTIONS Call-ID: 53216ecd6b19edcd59e4e6970fcbec78@192.168.10.225 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '53216ecd6b19edcd59e4e6970fcbec78@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '60dfba6a2ec5eebe307f598b6f321f77@192.168.10.225' Method: OPTIONS Really destroying SIP dialog '365b3715288e26f34c1c6fd9727cc1eb@192.168.10.225' Method: OPTIONS pbx*CLI> dGpbx*CLI> sip no debug SIP Debugging Disabled