[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

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>