[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

Ok thank you for your clear post, here is the grep of the call i made, it's a
outside call to asterisk then a transfer to opensips. (anounched) that one
is working, but then i try a transfer from 105 to 103, this is from opensips
extention to opensips extention. this one fails.

http://dl.getdropbox.com/u/1382962/grep.txt

Best regards


Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> Well yes, it does work for the internal calls, but
>> when a call comes in true asterisk to an opensips extention i CAN'T
>>  transfer it :-), i get transfer failed in my screen of my phone, and the
>>  call stays on the original called extention. This is only for announced
>>  transfers, unannounced works fine.
>> 
>> Flavio post stated something about routing your REFER's back to asterisk,
>>  so it should work.. but i don't know how to route these calls back to
>> the
>>  asterisk.
> 
> Please, you *already* have the answer. When a phone is speaking with
> Asterisk 
> (through OpenSIPS) you must route REFER to Asterisk as *any* other
> in-dialog 
> request, this is, the *same* as when a phone is speaking with other phone 
> directly (through OpenSIPS).
> 
> If the REFER fails this is because Asterisk is rejecting it !!!
> 
> I already suggested you to do a SIP capture (using ngrep) to inspect which 
> error replies Asterisk when the REFER arrives to it. Please do it and
> paste it 
> here (I expect a 403 or 404, so it means a wrong configuration in you 
> Asterisk, no more).
> 
> And please, forget anything about exotic routing of the REFER.
> 
> 
> -- 
> 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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