[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

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..



Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> Yes, i just noticed that error myself, that's something else didn't had
>>  that before today :-), but that's the whole issue, i think it's not
>>  sending a new refer, it just creates a new call on line 2, and when i
>> try
>>  to press transfer and hang up the call disappears and in my phone screen
>> i
>>  see "transfer failed"
> 
> 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.
> 
> 
>> 
>> the situation is like this: 104 is on Asterisk, 105 & 103 are on
>> opensips,
>> 104 takes all the outside calls (for now i made it like this, so we are
>>  able to transfer the calls announced)
>> 
>> i call from my mobile, true the sip trunk  to 104. I transfer a call from
>> 104 to 105, this works fine. Then i transfer the same call from 105 to
>> 103,
>> these last 2 are both opensips extensions..  and that last part, doesn't
>> work. the ngrep of a call like this is what you can see in my last post
> 
> It's not easy to guess the issue with this information. However, I
> describe a 
> flow that should work:
> 
> - Asterisk receives a call and calls to 104 (through OpenSIPS).
> - 104 transfers Asterisk to 105, so now Asterisk is speaking with 105.
> - 105 transfers Asterisk to 103, so now Asterisk is speaking with 103.
> 
> It should work if 105, 103 and 104 have the same configuration for
> OpenSIPS 
> and Asterisk.
> 
> 
>  
> 
> 
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
> 
> _______________________________________________
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> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

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