china*CLI> sip debug peer proxy01.sipphone.com SIP Debugging Enabled for IP: 198.65.166.131:5060 china*CLI> Sip read: INVITE sip:17476943246:@my.ip.goes.here SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 Record-Route: To: From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 CSeq: 1 INVITE Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 Content-Type: application/sdp Content-Length: 254 Contact: User-Agent: Callwave SipProxy v2.0 Max-Forwards: 16 RemoteIP: 64.211.149.50 P-hint: usrloc routed v=0 o=- 1346543798 1346543798 IN IP4 64.211.149.236 s=- t=0 0 m=audio 6192 RTP/AVP 18 0 c=IN IP4 64.211.149.236 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 m=audio 3008 RTP/AVP 96 c=IN IP4 64.211.149.50 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 17 headers, 12 lines Using latest request as basis request Sending to 198.65.166.131 : 5060 (non-NAT) Found peer 'proxy01.sipphone.com' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 96 Peer audio RTP is at port 64.211.149.236:3008 Found description format G729 Found description format PCMU Found description format iLBC Capabilities: us - 0x406 (gsm|ulaw|ilbc), peer - audio=0x504 (ulaw|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 17476943246: in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 To: ;tag=as5b30273a Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 198.65.166.131:5060 china*CLI> Sip read: ACK sip:17476943246:@my.ip.goes.here SIP/2.0 Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0 From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 To: ;tag=as5b30273a CSeq: 1 ACK Content-Length: 0 7 headers, 0 lines Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' china*CLI> Sip read: CANCEL sip:17476943246:@my.ip.goes.here SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 Record-Route: To: From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 CSeq: 1 CANCEL Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 Content-Length: 0 Max-Forwards: 16 P-hint: usrloc routed 13 headers, 0 lines Sending to 198.65.166.131 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 481 Call Leg Does Not Exist Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 To: ;tag=as74a51b73 Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 198.65.166.131:5060 Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' china*CLI> Sip read: CANCEL sip:17476943246:@my.ip.goes.here SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 Record-Route: To: From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 CSeq: 1 CANCEL Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 Content-Length: 0 Max-Forwards: 16 P-hint: usrloc routed 13 headers, 0 lines Sending to 198.65.166.131 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 481 Call Leg Does Not Exist Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 From: ;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 To: ;tag=as7ca411da Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 198.65.166.131:5060 Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' china*CLI> china*CLI> 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:198.65.166.131 SIP/2.0 Via: SIP/2.0/UDP my.ip.goes.here:5060;branch=z9hG4bK508fd4f4 From: "asterisk" ;tag=as7b232ec8 To: Contact: Call-ID: 042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 25 Aug 2006 17:25:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 198.65.166.131:5060 china*CLI> Sip read: SIP/2.0 404 Invalid number or user is offline Via: SIP/2.0/UDP my.ip.goes.here:5060;branch=z9hG4bK508fd4f4 From: "asterisk" ;tag=as7b232ec8 To: ;tag=21a483426c2cd5d9b85bffe6bba40a2e.8257 Call-ID: 042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here'