sipdebug2

Sip read:INVITE sip:192.168.0.1 SIP/2.0Content-Length: 173Contact: <sip:192.168.0.5:5060>Call-ID: 425C9413-6C0D-41F6-A214-5467DA42DA33@192.168.0.5Content-Type: application/sdpFrom: "test"<sip:192.168.0.5>;tag=343692083CSeq: 1 INVITEMax-Forwards: 70To: <sip:192.168.0.1>Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bKc0a800050131c9b1410cd942000009a900000031v=0o=- 3300349891 3300349891 IN IP4 192.168.0.5s=-c=IN IP4 192.168.0.5t=0 0m=audio 16416 RTP/AVP 0 3 8 101a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,1610 headers, 8 linesUsing latest request as basis requestSending to 192.168.0.5 : 5060 (non-NAT)Found audio format UNKNFound audio format UNKNFound audio format ALAWFound audio format UNKNFound description format telephone-eventCapabilities: us - 4, them - 14/0, combined - 4Non-codec capabilities: us - 1, them - 1, combined - 1Looking for 192.168.0.1 in sipTransmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bKc0a800050131c9b1410cd942000009a900000031From: "test"<sip:192.168.0.5>;tag=343692083To: <sip:192.168.0.1>;tag=as524a4322Call-ID: 425C9413-6C0D-41F6-A214-5467DA42DA33@192.168.0.5CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: <sip:192.168.0.1@192.168.0.1>Content-Length: 0 to 192.168.0.5:5060Sip read:ACK sip:192.168.0.1 SIP/2.0Content-Length: 0Call-ID: 425C9413-6C0D-41F6-A214-5467DA42DA33@192.168.0.5Max-Forwards: 70CSeq: 1 ACKFrom: "test"<sip:192.168.0.5>;tag=343692083To: <sip:192.168.0.1>;tag=as524a4322Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bKc0a800050131c9b1410cd942000009a9000000318 headers, 0 lines