[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
> ok so you mean it like this?
> 
> sip trunk -> opensips -> asterisk
> every call goes true opensips true asterisk to a extention, so asterisk
>  keep track of all the calls.
> 
> so when extention 085* comes in (outside number) i do a dial from opensips
> to asterisk, asterisk knows it should dial 105, and then i can transfer the
> 105 call to 103? I think this will work also.. 
> 
> One question tho, what do you mean with:
> 
> "It should work if 105, 103 and 104 have the same configuration for
>  OpenSIPS  and Asterisk."
> 
> I don't have any asterisk information in the phones now, they all registred
> on opensips..
> 

I don't fully understand your scenario. Please describe it to me as I 
described this one:

> This is how attended transfer works:
> - A is speaking with B.
> - A puts B on hold and sends a new INVITE to C (and talks with him).
> - A sends a REFER to B with "Refer-To: sip:C at domain;replaces=xxxx".
> - B then generates an INVITE to C with "Replaces" header.
> - C accepts the call and replaces the previous call (established with A) 
> since the new INVITE contains a "Replaces" header with previous dialog 
> information.

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