[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
> 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.

Please, you *already* have the answer. When a phone is speaking with Asterisk 
(through OpenSIPS) you must route REFER to Asterisk as *any* other in-dialog 
request, this is, the *same* as when a phone is speaking with other phone 
directly (through OpenSIPS).

If the REFER fails this is because Asterisk is rejecting it !!!

I already suggested you to do a SIP capture (using ngrep) to inspect which 
error replies Asterisk when the REFER arrives to it. Please do it and paste it 
here (I expect a 403 or 404, so it means a wrong configuration in you 
Asterisk, no more).

And please, forget anything about exotic routing of the REFER.


-- 
Iñaki Baz Castillo <ibc at aliax.net>