[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

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" 

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


Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> 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
> 
> I see no REFER request in that trace.
> Also what I see is a "484 Addres Incomplete" from Asterisk
> 
>> 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>
> 
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> 

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