[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

ok, but i want to have my users registered to opensips. And what happend
before was this ->
call from outside -> asterisk did a dial to 101 in opensips -> 101 ringed i
had a call..

transfer this call from 101 to 102 -> i've got a new call on line two from
phone 101 to 102, but the call from the outside, was on hold on 101 on line
1..  Blind transfers work perfectly, but not the announced transfers. they
just go on hold, and when i press transfer after telling 102 that i have a
call from 101, the call just stays on 101 in hold.

How did you fix this? Do you have working announced transfers?


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

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