<div>Hi users,</div>
<div> </div>
<div> This is ngrep report regarding my problem;</div>
<div> </div>
<div>-----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>-------------------------</div>
<div>U <a href="http://82.102.69.105:39871">82.102.69.105:39871</a> -> <a href="http://81.21.33.35:5060">81.21.33.35:5060</a><br>INVITE sip:99106883@81.21.33.35:5060 SIP/2.0.<br>To: "99106883"<sip:99106883@81.21.33.35
:5060>.<br>From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8.<br>Via: SIP/2.0/UDP <a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport.
<br>Via: SIP/2.0/UDP <a href="http://192.168.1.102">192.168.1.102</a>;branch=z9hG4bKf638e18b56022ea3.<br>Call-ID: <a href="mailto:a78d5c993a9dd6b4@192.168.1.102">a78d5c993a9dd6b4@192.168.1.102</a>.<br>CSeq: 47344 INVITE.<br>
Record-Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>Contact: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>Max-Forwards: 69.<br>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
<br>Content-Type: application/sdp.<br>Supported: replaces.<br>User-Agent: Grandstream BT110 <a href="http://1.0.8.23">1.0.8.23</a>.<br>Content-Length: 361.<br>.<br>v=0.<br>o=line2 8000 8000 IN IP4 <a href="http://192.168.1.102">
192.168.1.102</a>.<br>s=SIP Call.<br>c=IN IP4<font color="#3366ff"> </font><font color="#ff0000">82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy ,<br></font>t=0 0.<br>m=audio 5004 RTP/AVP 18 4 2 97 9 0 101.
<br>a=fmtp:97 mode=20.<br>a=fmtp:101 0-11.<br>a=ptime:20.<br>a=rtpmap:18 G729/8000.<br>a=rtpmap:4 G723/8000.<br>a=rtpmap:2 G726-32/8000.<br>a=rtpmap:97 iLBC/8000.<br>a=rtpmap:9 G722/16000.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:101 telephone-event/8000.
<br>a=sendrecv.<br> </div>
<div>-----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"-----------------------------------</div>
<div> </div>
<div>U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> pstngw:5060<br>INVITE sip:99106883@pstngw:5060 SIP/2.0.<br>Record-Route: <sip:99106883@81.21.33.35:5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>.<br>To: "99106883"<
sip:99106883@81.21.33.35:5060>.<br>From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8.<br>Via: SIP/2.0/UDP <a href="http://81.21.33.35">81.21.33.35</a>;branch=z9hG4bK0ab.9522bc25.0.<br>Via: SIP/2.0/UDP
<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;received=<a href="http://82.102.69.105">82.102.69.105</a>;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871.<br>Via: SIP/2.0/UDP <a href="http://192.168.1.102">
192.168.1.102</a>;branch=z9hG4bKf638e18b56022ea3.<br>Call-ID: <a href="mailto:a78d5c993a9dd6b4@192.168.1.102">a78d5c993a9dd6b4@192.168.1.102</a>.<br>CSeq: 47344 INVITE.<br>Record-Route: <sip:<a href="http://192.168.1.100:5060">
192.168.1.100:5060</a>>.<br>Contact: <sip:<a href="http://82.102.69.105:39871">82.102.69.105:39871</a>>.<br>Max-Forwards: 16.<br>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.<br>Content-Type: application/sdp.
<br>Supported: replaces.<br>User-Agent: Grandstream BT110 <a href="http://1.0.8.23">1.0.8.23</a>.<br>Content-Length: 360.<br>.<br>v=0.<br>o=line2 8000 8000 IN IP4 <a href="http://192.168.1.102">192.168.1.102</a>.<br>s=SIP Call.
<br>c=IN IP4<font color="#ff99ff"> </font><font color="#ff6666">81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw)</font><br>t=0 0.<br>m=audio 60516 RTP/AVP 18 4 2 97 9 0 101.
<br>a=fmtp:97 mode=20.<br>a=fmtp:101 0-11.<br>a=ptime:20.<br>a=rtpmap:18 G729/8000.<br>a=rtpmap:4 G723/8000.<br>a=rtpmap:2 G726-32/8000.<br>a=rtpmap:97 iLBC/8000.<br>a=rtpmap:9 G722/16000.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:101 telephone-event/8000.
<br>a=sendrecv.<br> </div>
<div>---------------------------------------------<and the pstngw is making call>-----------------------------------------------</div>
<div> </div>
<div>U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> pstngw:39871<br>SIP/2.0 183 Session Progress.<br>Via: SIP/2.0/UDP <a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;received=<a href="http://82.102.69.105">
82.102.69.105</a>;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP <a href="http://192.168.1.102">192.168.1.102</a>;branch=z9hG4bK01c64e769ba6c176.<br>From: "12345"<sip:12345@81.21.33.35
:5060>;tag=9ca5c1fe3b43aad6.<br>To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764.<br>Date: Wed, 06 Sep 2006 10:27:40 GMT.<br>Call-ID: <a href="mailto:72d0e4e8adca2ab2@192.168.1.102">72d0e4e8adca2ab2@192.168.1.102
</a>.<br>Server: Cisco-SIPGateway/IOS-12.x.<br>CSeq: 32352 INVITE.<br>Allow-Events: telephone-event.<br>Content-Type: application/sdp.<br>Content-Disposition: session;handling=required.<br>Content-Length: 237.<br>.<br>v=0.
<br>o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw<br>s=SIP Call.<br>c=IN IP4 <font color="#ff6666">81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>---------------------
</font><br>t=0 0.<br>m=audio 60518 RTP/AVP 18 100.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=yes.<br>a=rtpmap:100 X-NSE/8000.<br>a=fmtp:100 192-194.<br>a=ptime:20.<br> </div>
<div>---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this----------------</div>
<div> </div>
<div>
<p>U pstngw:52991 -> <a href="http://81.21.33.35:5060">81.21.33.35:5060</a><br>BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0.<br>Via: SIP/2.0/UDP <a href="http://81.21.38.15:5060">81.21.38.15:5060
</a>.<br>From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292.<br>To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8.<br>Date: Wed, 06 Sep 2006 10:35:07 GMT.<br>Call-ID:
<a href="mailto:c2dd3fb9554ef6e4@192.168.1.102">c2dd3fb9554ef6e4@192.168.1.102</a>.<br>User-Agent: Cisco-SIPGateway/IOS-12.x.<br>Max-Forwards: 6.<br>Route: <font color="#ff0000"><sip:<a href="http://192.168.1.100:5060">
192.168.1.100:5060</a>>, <sip:<a href="http://82.102.69.105:39871">82.102.69.105:39871</a>>.------------------------> What is happening here ???????<br></font>Timestamp: 1157538919.<br>CSeq: 101 BYE.<br>Content-Length: 0.
<br>.</p>
<p>#<br>U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> <font color="#ff6666">192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use <font color="#ff0000">
<a href="http://82.102.69.105">82.102.69.105</a> </font></font><br>BYE sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a> SIP/2.0.<br>Record-Route: <sip:<a href="http://81.21.33.35">81.21.33.35</a>;ftag=7E66E3D8-1292;lr=on>.
<br>Via: SIP/2.0/UDP <a href="http://81.21.33.35">81.21.33.35</a>;branch=z9hG4bKb0f1.3fe32691.0.<br>Via: SIP/2.0/UDP pstngw:5060.<br>From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292.<br>To: "12345"<
sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8.<br>Date: Wed, 06 Sep 2006 10:35:07 GMT.<br>Call-ID: <a href="mailto:c2dd3fb9554ef6e4@192.168.1.102">c2dd3fb9554ef6e4@192.168.1.102</a>.<br>User-Agent: Cisco-SIPGateway/IOS-
12.x.<br>Max-Forwards: 5.<br>Route: <sip:<a href="http://82.102.69.105:39871">82.102.69.105:39871</a>>.<br>Timestamp: 1157538919.<br>CSeq: 101 BYE.<br>Content-Length: 0.<br>P-hint: rr-enforced.<br></p></div>
<div>So the call never ends up from SIP SERVER to other party</div>
<div> </div>
<div> Make some comments on above and assist me where am I going wrong:</div>
<div> </div>
<div>and i am using <a href="http://www.openser.org">www.openser.org</a> pstn default script as ser.cfg</div>
<div> </div>
<div> Hope to get some help </div>
<div> Thanks,</div>
<div>Ravi.</div>