[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

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..

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


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>
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

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