[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

Well, it doesn't matter if i hang up, use the transfer button again or
anything the result is the same, my call is gone. :)

I understand the rest of your post tho, :) thanks alot for your input.



Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> Ok well:
>> 
>> A = mobile phone
>> B = 101
>> C = 102
>> 
>> A calls B, true the sip trunk -> connected to opensips -> that dials B
>> true
>> Asterisk (B is connected to opensips).
> 
> so:
> 
>   A -- Provider -- OpenSIPS -- Asterisk -- B
>   A -- Provider -- OpenSIPS -- C
> 
> 
>> B transfers to C:
> 
>> B puts A on hold
> 
> No, that's an error. B puts *Asterisk* on hold, but not A (you cannot see
> a 
> re-INVITE from Asterisk to A, but just from B to Asterisk).
> 
> 
>> -> dials C true Asterisk and opensips (C is also connected
>> to opensips) there is a call from B to C now,
> 
> No, there are these calls:
> 
> Call between B and A (two dialogs in fact):
> - B <--> Asterisk
> - Asterisk <--(opensips)--> <--(provider)--> A
> 
> Call between B anc C (two dialogs in fact):
> - B <--> Asterisk
> - Asterisk <--(opensips)--> C
> 
> 
>> B tells C that A is there,
>>  and B hangs up.
> 
> Ohhh, "B hangs up"... so you are doing Asterisk transfer instead of SIP 
> transfer??? (so you are doing the transference using DTFM???)
> 
> The of course there is NO REFER !
> 
> However it doesn't affect the scenario.
> 
> 
> 
>>  The call goes from B to C, the call is now connected true
>>  the A <-> sip trunk <-> opensips <-> asterisk <-> opensips <-> C.
> 
> 
>> I don't see another way to do this.
> 
> Yes, it's valid.
> 
> 
> 
> 
> 
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
> 
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> 
> 

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