31 Jan
2008
31 Jan
'08
3:08 p.m.
El Jueves, 31 de Enero de 2008, Jan ONDREJ (SAL) escribió:
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
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@IP_A
Refer-To: sip:C@domain
- So A will receive that REFER and then A will generate an INVITE to C:
INVITE sip:C@domain
From: A@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.