Could you explain which devices have what IP and what is behind NAT between what?<br><br><div><span class="gmail_quote">On 12/4/06, <b class="gmail_sendername">Matt</b> <<a href="mailto:mhoppes@gmail.com">mhoppes@gmail.com
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Debug of the sip peer 126 shows:<br><br> -- Called 126<br> -- Agent/9999 is ringing
<br>Retransmitting #1 (NAT) to <a href="http://63.174.244.196:5060">63.174.244.196:5060</a>:<br>INVITE <a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://63.174.244.175:5060">
63.174.244.175:5060</a>;branch=z9hG4bK598573e4;rport<br>From: "Test VoIP Accounts"" <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>>;tag=as1a3a38f5<br>To: <<a href="mailto:sip:126@63.174.244.196">
sip:126@63.174.244.196</a>><br>Contact: <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>><br>Call-ID: <a href="mailto:1882bae616cf60be14d2436e73c7026a@63.174.244.175">1882bae616cf60be14d2436e73c7026a@63.174.244.175
</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 04 Dec 2006 20:42:26 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 275
<br><br>v=0<br>o=root 3555 3555 IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>s=session<br>c=IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>t=0 0<br>m=audio 19720 RTP/AVP 0 97 111 101<br>a=rtpmap:0 PCMU/8000
<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:111 G726-32/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---<br>Retransmitting #2 (NAT) to <a href="http://63.174.244.196:5060">63.174.244.196:5060
</a>:<br>INVITE <a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://63.174.244.175:5060">63.174.244.175:5060</a>;branch=z9hG4bK598573e4;rport<br>From: "Test VoIP Accounts"" <
<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>>;tag=as1a3a38f5<br>To: <<a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a>><br>Contact: <<a href="mailto:sip:5706016716@63.174.244.175">
sip:5706016716@63.174.244.175</a>><br>Call-ID: <a href="mailto:1882bae616cf60be14d2436e73c7026a@63.174.244.175">1882bae616cf60be14d2436e73c7026a@63.174.244.175</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70
<br>Date: Mon, 04 Dec 2006 20:42:26 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 275<br><br>v=0<br>o=root 3555 3555 IN IP4 <a href="http://63.174.244.175">
63.174.244.175</a><br>s=session<br>c=IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>t=0 0<br>m=audio 19720 RTP/AVP 0 97 111 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:111 G726-32/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---<br>Retransmitting #3 (NAT) to <a href="http://63.174.244.196:5060">63.174.244.196:5060</a>:<br>INVITE <a href="mailto:sip:126@63.174.244.196">
sip:126@63.174.244.196</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://63.174.244.175:5060">63.174.244.175:5060</a>;branch=z9hG4bK598573e4;rport<br>From: "Test VoIP Accounts"" <<a href="mailto:sip:5706016716@63.174.244.175">
sip:5706016716@63.174.244.175</a>>;tag=as1a3a38f5<br>To: <<a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a>><br>Contact: <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175
</a>><br>Call-ID: <a href="mailto:1882bae616cf60be14d2436e73c7026a@63.174.244.175">1882bae616cf60be14d2436e73c7026a@63.174.244.175</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 04 Dec 2006 20:42:26 GMT
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 275<br><br>v=0<br>o=root 3555 3555 IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>s=session
<br>c=IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>t=0 0<br>m=audio 19720 RTP/AVP 0 97 111 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:111 G726-32/8000<br>a=rtpmap:101 telephone-event/8000
<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---<br>Retransmitting #4 (NAT) to <a href="http://63.174.244.196:5060">63.174.244.196:5060</a>:<br>INVITE <a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196
</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://63.174.244.175:5060">63.174.244.175:5060</a>;branch=z9hG4bK598573e4;rport<br>From: "Test VoIP Accounts"" <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175
</a>>;tag=as1a3a38f5<br>To: <<a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a>><br>Contact: <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>><br>Call-ID: <a href="mailto:1882bae616cf60be14d2436e73c7026a@63.174.244.175">
1882bae616cf60be14d2436e73c7026a@63.174.244.175</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 04 Dec 2006 20:42:26 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Content-Type: application/sdp<br>Content-Length: 275<br><br>v=0<br>o=root 3555 3555 IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>s=session<br>c=IN IP4 <a href="http://63.174.244.175">63.174.244.175</a>
<br>t=0 0<br>m=audio 19720 RTP/AVP 0 97 111 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:111 G726-32/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---
<br>Retransmitting #5 (NAT) to <a href="http://63.174.244.196:5060">63.174.244.196:5060</a>:<br>INVITE <a href="mailto:sip:126@63.174.244.196">sip:126@63.174.244.196</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://63.174.244.175:5060">
63.174.244.175:5060</a>;branch=z9hG4bK598573e4;rport<br>From: "Test VoIP Accounts"" <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>>;tag=as1a3a38f5<br>To: <<a href="mailto:sip:126@63.174.244.196">
sip:126@63.174.244.196</a>><br>Contact: <<a href="mailto:sip:5706016716@63.174.244.175">sip:5706016716@63.174.244.175</a>><br>Call-ID: <a href="mailto:1882bae616cf60be14d2436e73c7026a@63.174.244.175">1882bae616cf60be14d2436e73c7026a@63.174.244.175
</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 04 Dec 2006 20:42:26 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Content-Type: application/sdp<br>Content-Length: 275
<br><br>v=0<br>o=root 3555 3555 IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>s=session<br>c=IN IP4 <a href="http://63.174.244.175">63.174.244.175</a><br>t=0 0<br>m=audio 19720 RTP/AVP 0 97 111 101<br>a=rtpmap:0 PCMU/8000
<br>a=rtpmap:97 iLBC/8000<br>a=rtpmap:111 G726-32/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>What is doing this?<br>_______________________________________________<br>
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