[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

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).

B transfers to C:
B puts A on hold -> dials C true Asterisk and opensips (C is also connected
to opensips) there is a call from B to C now, B tells C that A is there, and
B hangs up. 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.

What i have now is: 
sip trunk <-> Asterisk <-> opensips <-> Extensions and in this scenario  the
outside transfer true Asterisk fails. or atleast, there is no REFER send, i
mean the call DOES go on hold on B, and C is called, but when B disconnects
with C, (so this is the moment the call should actually transfer) the call
from A to B is still there on hold, and is never transferred.

Is this such a strange scenario? That the call never makes the final
destination? 



Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> ok so you mean it like this?
>> 
>> sip trunk -> opensips -> asterisk
>> every call goes true opensips true asterisk to a extention, so asterisk
>>  keep track of all the calls.
>> 
>> so when extention 085* comes in (outside number) i do a dial from
>> opensips
>> to asterisk, asterisk knows it should dial 105, and then i can transfer
>> the
>> 105 call to 103? I think this will work also.. 
>> 
>> One question tho, what do you mean with:
>> 
>> "It should work if 105, 103 and 104 have the same configuration for
>>  OpenSIPS  and Asterisk."
>> 
>> I don't have any asterisk information in the phones now, they all
>> registred
>> on opensips..
>> 
> 
> I don't fully understand your scenario. Please describe it to me as I 
> described this one:
> 
>> This is how attended transfer works:
>> - A is speaking with B.
>> - A puts B on hold and sends a new INVITE to C (and talks with him).
>> - A sends a REFER to B with "Refer-To: sip:C at domain;replaces=xxxx".
>> - B then generates an INVITE to C with "Replaces" header.
>> - C accepts the call and replaces the previous call (established with A) 
>> since the new INVITE contains a "Replaces" header with previous dialog 
>> information.
> 
> -- 
> 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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