<--- SIP read from UDP://72.187.243.97:45990 ---> INVITE sip:19544447408@66.28.190.219 SIP/2.0 Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK66E70C080F304B2DB834F28A64B4CA47 From: 17864335989 ;tag=3016316444 To: Contact: Call-ID: F1F160F9-4796-4E89-81A2-1182C550AE0E@192.168.1.111 CSeq: 19716 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-PRO release 1105x Content-Length: 218 v=0 o=17864335989 177936825 177937141 IN IP4 72.187.243.97 s=X-PRO c=IN IP4 72.187.243.97 t=0 0 m=audio 45992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (11 headers 10 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to 72.187.243.97 : 45990 (NAT) Using INVITE request as basis request - F1F160F9-4796-4E89-81A2-1182C550AE0E@192.168.1.111 No user '17864335989' in SIP users list Found peer '72.187.243.97' for '17864335989' from 72.187.243.97:45990 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 72.187.243.97:45992 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 72.187.243.97:45992 Looking for 19544447408 in inbound (domain 66.28.190.219) <--- Reliably Transmitting (NAT) to 72.187.243.97:45990 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 72.187.243.97:45990;branch=z9hG4bK66E70C080F304B2DB834F28A64B4CA47;received=72.187.243.97 From: 17864335989 ;tag=3016316444 To: ;tag=as70732a79 Call-ID: F1F160F9-4796-4E89-81A2-1182C550AE0E@192.168.1.111 CSeq: 19716 INVITE User-Agent: CiscoSystemsSIP-GW-UserAgent Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0