[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

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>
> 
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> 
> 

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