[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

BTW, when transfering anounched, i don't see a refer coming to asterisk, so
it stays in opensips..
If anybody wants i can add my script + a log file for this problem.


Peter den Hartog wrote:
> 
> Well yes, it does work for the internal calls, but
> when a call comes in true asterisk to an opensips extention i CAN'T
> transfer it :-), i get transfer failed in my screen of my phone, and the
> call stays on the original called extention. This is only for announced
> transfers, unannounced works fine. 
> 
> Flavio post stated something about routing your REFER's back to asterisk,
> so it should work.. but i don't know how to route these calls back to the
> asterisk.
> 
> 
> 
> Iñaki Baz Castillo wrote:
>> 
>> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>>>         if (is_method("REFER")) {
>>>         route(4);
>>>         }
>>> 
>>> And route(4) is the drouting script, so then it should go back to the
>>> gateway (asterisk) that knows it should do a dial to 103 right?
>> 
>> Not at all. REFER is an in-dialog request so leave it going through the 
>> "loose_route" secion, just it. It MUST work. 
>> 
>> 
>> -- 
>> Iñaki Baz Castillo <ibc at aliax.net>
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
> 
> 

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