Hi Nauman,

Just a wild guess - maybe VoipFone sends CANCEL after the second 180 
because of different to-tags in first 180 (from UAC1) and the second 180 
(from UAC2) - normally it should not be a problem, but you never know 
with all these implementations...

Regarding sipgate, it may be because of the same cause.....

We can try a simple experiment to strip out the to-tag from all 
provisional replies - to see if it works or not.

Regards,
Bogdan

Nauman Sulaiman wrote:
> Hi , we have a set up where we are using a SIP proxy( OpenSIPS) to handle a 
> 302 redirect locally but it doesn't seem to be working with voip providers 
> here is the call flow. Is it legal, if so we'll bother you with a SIP trace. 
> The 302 is handled locally by the OpenSIPS proxy and generates a new call leg 
> but can't establish a dialog with Sipgate or Voipfone
>
>                  INVITE             INVITE
> VoipFone Server ---------> Proxy ------------> UAC1
>                   Trying
> VoipFone  Server <--------  Proxy
>                                     180 Ring
>                  180 Ring       <------------- UAC1
>                 <------    Proxy    302 Redir
>                                 <-------------- UAC1
>                                     INVITE
>                                 ---------------> UAC2
>                                    180 Ring
>                                 <----------------UAC2
>                   180 Ring
> VoipFone Server <----------   
>                   CANCEL     
>                ------------>         200 OK
>                  200 OK         <--------------- UAC2
>                <------------
>
> THe VoipFone server sends a cancel straight after the 180 ring, Sipgate
> doesn't do this however it never sends an ACK for the 200K, anyway both
> don't like what we are doing.
>
> We need to do the 302 redirect locally on our proxy as not all Voip providers 
> support it so we can't let it go all the way back. Hope its clear what we are 
> trying to, and is there any way forward.
>
> Here is the code in our opensips.cfg file 
>
> failure_route[1] {
>         if (t_was_cancelled()) {
>                 exit;
>         }
>
>         get_redirects("3:1");
>         t_relay();
> }
>
> Thanks
>
>
>
>       
>
> _______________________________________________
> Users mailing list
> [email protected]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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