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<p>HI Nauman,<br>
<br>
Just a guess, but does the ACK have a different R-URI than the reINVITE (as sent by the client)? I ask because the 200 OK to the reINVITE carries a contact address which is used to update the peer address of the dialog state, which shows up as the R-URI of the ACK.<br>
<br>
Regards,<br>
Kennard<br>
<br>
<img width="16" height="16" src="cid:1__=07BBFD3DDFC41F5F8f9e8a93df9@logitech.com" border="0" alt="Inactive hide details for Bogdan-Andrei Iancu ---09/30/2010 12:30:14 AM---Hi Nauman, Do you get any errors in log regarding TLS"><font color="#424282">Bogdan-Andrei Iancu ---09/30/2010 12:30:14 AM---Hi Nauman, Do you get any errors in log regarding TLS connection or others?</font><br>
<br>
<font size="2" color="#5F5F5F">From: </font><font size="2">Bogdan-Andrei Iancu <bogdan@voice-system.ro></font><br>
<font size="2" color="#5F5F5F">To: </font><font size="2">nauman762-home@yahoo.co.uk, OpenSIPS users mailling list <users@lists.opensips.org></font><br>
<font size="2" color="#5F5F5F">Date: </font><font size="2">09/30/2010 12:30 AM</font><br>
<font size="2" color="#5F5F5F">Subject: </font><font size="2">Re: [OpenSIPS-Users] Problem with Opensips routing Reinvite ACK</font><br>
<font size="2" color="#5F5F5F">Sent by: </font><font size="2">users-bounces@lists.opensips.org</font><br>
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<tt>Hi Nauman,<br>
<br>
Do you get any errors in log regarding TLS connection or others?<br>
<br>
Also, after loose_route() print via xlog the $ru and $du to see where <br>
opensips tries to route the ACK.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
Nauman Sulaiman wrote:<br>
> Hi, we have a set up as follows using Opensips 1.6.2<br>
><br>
> UA -> OpenSIPS -> SIP Provider<br>
><br>
> TLS TLS UDP<br>
> UDP<br>
><br>
> We can set up a call from Sip Provider to UA. However when we issue a Reinvite due to (Hold) from UA the Sip provider gets the invite and sends 200K to which the UA responds with an ACK , however Opensips does not seem<br>
> to be routing this ACK back to the provider. Strangely the Reinvite which initally started this has the same route set, yet that reaches the provider.<br>
><br>
> ACK sip:anonymous@217.10.79.23:5060 SIP/2.0<br>
> Via: SIP/2.0/TLS 192.168.0.4;branch=z9hG4bKyQatKQmaHK0Br<br>
> Route:<br>
> <sip:178.230.138.190:5061;transport=tls;r2=on;lr=on;ftag=as7615d7df><br>
> Route: <sip:178.230.138.190;r2=on;lr=on;ftag=as7615d7df><br>
> Route: <sip:217.10.79.23;lr=on;ftag=as7615d7df><br>
> Route: <sip:172.20.40.3;lr=on><br>
> Route: <sip:217.10.79.23;lr=on;ftag=as7615d7df><br>
> Max-Forwards: 70<br>
> From: <sip:00441619082575@sipgate.co.uk>;tag=t8B2BFQDUgNvc<br>
> To: "anonymous" <sip:anonymous@sipgate.co.uk>;tag=as7615d7df<br>
> Call-ID: 27a6f665224ffd7b7c5e0d2616be4dee@sipgate.co.uk<br>
> CSeq: 2556588 ACK<br>
> Content-Length: 0<br>
><br>
><br>
><br>
><br>
><br>
><br>
> <br>
><br>
> _______________________________________________<br>
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><br>
> <br>
<br>
<br>
-- <br>
Bogdan-Andrei Iancu<br>
OpenSIPS Bootcamp<br>
15 - 19 November 2010, Edison, New Jersey, USA<br>
</tt><tt>www.voice-system.ro</tt><tt><br>
<br>
<br>
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