[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

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>