[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

El Viernes, 23 de Octubre de 2009, Peter den Hartog escribió:
> Oke that is clear to me,
> 
> So if i want the transfer function, i have to use a asterisk as a gateway
> (or any other pbx).
> But how do you create this new channel? i tried this before, with an
> asterisk as pbx, that received the outside calls... i've created a dial
>  from asterisk to my opensips extention, but this is clearly wrong because
>  if i try a transfer then, he tries it on the asterisk.. result the new
> destination not found..

If "destination not found" that means that your dialplan is not correctly 
configured. If user1 transfer to Asterisk to user2, this means that Asterisk 
will look for "user2" extension in the context of OpenSIPs peer. So "exten => 
user2" must exist there.


> What i understand from your story is that asterisk shouldn't do a dial, but
> a invite to the opensips extention, am i right?

No, it's the same "doing a dial" and "sending an invite". Asterisk just can 
send an INVITE.


> Any ideas on how to do this?

There are some doucments/howtos in OpenSIPS wiki about integration woth 
Asterisk. Not sure if they fully cover users integration (so transfer is 
possible and so).
However I've this exact scenario working perfectly. It's just a taste of 
configuration (peers and dialplan in Asterisk).




> 
> Iñaki Baz Castillo wrote:
> > El Viernes, 23 de Octubre de 2009, Peter den Hartog escribió:
> >> yeah i understand that, but why is it sending this refer to the sip
> >> trunk?
> >>  i mean..
> >>
> >> it's an outside call going to a local extention, i want to transfer from
> >> 1
> >> local extention to another, so why isn't my opensips doing this refer?
> >
> > Sorry but you don't seem to understand how a REFER transfer works:
> >
> > 1) Your PSTN provider sends an INVITE to your proxy (opensips).
> > 2) OpenSIPS routes the call to user1.
> > 3) User1 answers the call and so.
> > 4) User1 wants to transfer the call to user2.
> > 5) User1 sends a REFER to *the PSTN provider* (through OpenSIPS as any
> > in- dialog request).
> > 6) The PSTN provider accepts the refer so *initiates* a new call to
> > user2.
> >
> > Of course point 5 will NEVER work with a PSTN provider (and shouldn't
> > work at
> > all!). This is why PBX/B2BUA do exist: to enable PBX features.
> >
> > If you just have a proxy and receive calls from a PSTN you could NEVER
> > transfer that call to other user.
> >
> > Having a PBX/B2BUA the sceario changes and allows transference. An
> > example scenario:
> >
> > 1) Your PSTN provider sends an INVITE to your PBX (B2BUA).
> > 2) The PBX generates a *new* INVITE (a different dialog) to OpenSIPS.
> > 3) OpenSIPS routes the call to user1.
> > 4) User1 answers the call.
> > 5) User1 wants to transfer the call to user2.
> > 6) User1 sends a REFER to *the PBX* (through OpenSIPS as any in-dialog
> > request).
> > 7) The PBX  provider accepts the refer so *initiates* a new call to
> > user2. 8) The PSTN provider didn't realize, at all, about the
> > transference.
> 


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