pbx*CLI> pbx*CLI> Reliably Transmitting (no NAT) to 66.51.127.173:5060: OPTIONS sip:sip.ca1.link2voip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.170:5070;branch=z9hG4bK460bc1fb;rport Max-Forwards: 70 From: "asterisk" ;tag=as466d2b34 To: Contact: Call-ID: 7c2324843d7e355e3a32f7dd59e84a10@192.168.2.170 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 11 Mar 2010 02:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (no NAT) to 66.51.110.210:5060: OPTIONS sip:sip.ca2.link2voip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.170:5070;branch=z9hG4bK2210ec6d;rport Max-Forwards: 70 From: "asterisk" ;tag=as31c5e711 To: Contact: Call-ID: 7d2113d50212a0ca29b83a144869409e@192.168.2.170 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 11 Mar 2010 02:34:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- pbx*CLI> <--- SIP read from UDP://66.51.127.173:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.170:5070;branch=z9hG4bK460bc1fb;rport=1024;received=99.240.223.105 From: "asterisk" ;tag=as466d2b34 To: ;tag=6da5cb3c58ecfc1b91772f44357856fa.87b2 Call-ID: 7c2324843d7e355e3a32f7dd59e84a10@192.168.2.170 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '7c2324843d7e355e3a32f7dd59e84a10@192.168.2.170' Method: OPTIONS pbx*CLI> <--- SIP read from UDP://66.51.110.210:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.170:5070;branch=z9hG4bK2210ec6d;rport=1024;received=99.240.223.105 From: "asterisk" ;tag=as31c5e711 To: ;tag=7a3986283a1d243977b41418b099c23d.e75f Call-ID: 7d2113d50212a0ca29b83a144869409e@192.168.2.170 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '7d2113d50212a0ca29b83a144869409e@192.168.2.170' Method: OPTIONS pbx*CLI> <--- SIP read from UDP://66.51.110.210:5060 ---> INVITE sip:16132885759@192.168.2.170:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK410d.7bde7436.0 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKNeHg7t6m61SFa Max-Forwards: 66 From: "JOULE MEDIA" ;tag=rD6KeyN2QS0Bc To: Call-ID: 6576de17-a759-122d-4aa5-b31d4caf95a3 CSeq: 128019085 INVITE Contact: User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 324 Remote-Party-ID: "JOULE MEDIA" ;screen=yes;privacy=off v=0 o=CiscoSystemsSIP-GW-UserAgent 3916750531371826529 8552236689903915812 IN IP4 66.51.127.163 s=SIP Call c=IN IP4 66.51.110.210 t=0 0 m=audio 15692 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=nortpproxy:yes <-------------> --- (18 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 66.51.110.210 : 5060 (no NAT) Using INVITE request as basis request - 6576de17-a759-122d-4aa5-b31d4caf95a3 Found peer '6132885759-sw2' for '6135651370' from 66.51.110.210:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Found audio description format CN for ID 13 Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.51.110.210:15692 Looking for 16132885759 in incoming-bogus-calls (domain 192.168.2.170) <--- Reliably Transmitting (no NAT) to 66.51.110.210:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK410d.7bde7436.0;received=66.51.110.210 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKNeHg7t6m61SFa From: "JOULE MEDIA" ;tag=rD6KeyN2QS0Bc To: ;tag=as089527d7 Call-ID: 6576de17-a759-122d-4aa5-b31d4caf95a3 CSeq: 128019085 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6576de17-a759-122d-4aa5-b31d4caf95a3' in 6400 ms (Method: INVITE) pbx*CLI> <--- SIP read from UDP://66.51.110.210:5060 ---> ACK sip:16132885759@192.168.2.170:5070 SIP/2.0 Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK410d.7bde7436.0 From: "JOULE MEDIA" ;tag=rD6KeyN2QS0Bc Call-ID: 6576de17-a759-122d-4aa5-b31d4caf95a3 To: ;tag=as089527d7 CSeq: 128019085 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '6576de17-a759-122d-4aa5-b31d4caf95a3' Method: ACK pbx*CLI> <--- SIP read from UDP://66.51.127.173:5060 ---> INVITE sip:16132885759@192.168.2.170:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK781f.99b221c4.0 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpQa98NQr3ag2N Max-Forwards: 66 From: "JOULE MEDIA" ;tag=SpZcgS65m2pyQ To: Call-ID: 658660e7-a759-122d-4aa5-b31d4caf95a3 CSeq: 128019085 INVITE Contact: User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 324 Remote-Party-ID: "JOULE MEDIA" ;screen=yes;privacy=off v=0 o=CiscoSystemsSIP-GW-UserAgent 6052559648260482984 9112528946230064074 IN IP4 66.51.127.163 s=SIP Call c=IN IP4 66.51.127.173 t=0 0 m=audio 14000 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=nortpproxy:yes <-------------> --- (18 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 66.51.127.173 : 5060 (no NAT) Using INVITE request as basis request - 658660e7-a759-122d-4aa5-b31d4caf95a3 Found peer '6132885759-sw1' for '6135651370' from 66.51.127.173:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Found audio description format CN for ID 13 Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.51.127.173:14000 Looking for 16132885759 in incoming-bogus-calls (domain 192.168.2.170) <--- Reliably Transmitting (no NAT) to 66.51.127.173:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK781f.99b221c4.0;received=66.51.127.173 Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpQa98NQr3ag2N From: "JOULE MEDIA" ;tag=SpZcgS65m2pyQ To: ;tag=as459696b1 Call-ID: 658660e7-a759-122d-4aa5-b31d4caf95a3 CSeq: 128019085 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '658660e7-a759-122d-4aa5-b31d4caf95a3' in 6400 ms (Method: INVITE) pbx*CLI> <--- SIP read from UDP://66.51.127.173:5060 ---> ACK sip:16132885759@192.168.2.170:5070 SIP/2.0 Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK781f.99b221c4.0 From: "JOULE MEDIA" ;tag=SpZcgS65m2pyQ Call-ID: 658660e7-a759-122d-4aa5-b31d4caf95a3 To: ;tag=as459696b1 CSeq: 128019085 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '658660e7-a759-122d-4aa5-b31d4caf95a3' Method: ACK pbx*CLI>