[OpenSER-Users] Call transfer
Iñaki Baz Castillo
ibc at aliax.net
Thu Jan 31 16:08:04 CET 2008
El Jueves, 31 de Enero de 2008, Jan ONDREJ (SAL) escribió:
> > case 1)
> >
> > 3 phones (A, B, C)
> >
> > - A calls B and they speak.
> >
> > - B wants to transfer the call to C. So B sends a REFER to A like this:
> > REFER sip:A at IP_A
> > Refer-To: sip:C at domain
> >
> > - So A will receive that REFER and then A will generate an INVITE to C:
> > INVITE sip:C at domain
> > From: A at domain
> >
> > - So in conclusion: a REFER in fact means: "please, invite this other
> > user".
>
> I am trying call transfer in this situation (case 1). It is not working in
> my default cofnig (attached to this email, there are just removed
> authentications to simply my testing).
> Tryed blind transfer and also managed transfer.
> One of three phnes still goes to undefined state (for example holding an
> nonexistent call). In different situations different phones.
Maybe that is a phone issue. I can sure that SIP transfer works perfectly
between some UAC's as Twinkle, Linksys SPA, Thomson S2030...
Of course, nothing special in OpenSer, it just has to route an in-dialog
REFER, no more.
Try doing a SIP trace in OpenSer host with ngrep:
ngrep -d any -P ' ' -W byline -T -t "" port 5060
And examinate it.
--
Iñaki Baz Castillo
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