[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

Well in that case, phone 1 + 2 are in opensips. and phone 1 has a call with a
mobile phone true asterisk.


Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> BTW, when transfering anounched, i don't see a refer coming to asterisk,
>> so
>> it stays in opensips..
> 
> Please describe your escenario:
> 
> If is the following?
> 
> - phone_1 speaking to Asterisk through OpenSIPS.
> - Asterisk speaking to phone_2 through OpenSIPS.
> 
> 
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
> 
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> 
> 

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