[Asterisk-Dev] Transferring Calls

[Michael Procter]

From: Christopher Jacob [mailto:chris at jacob-solutions.com]
> 
> I don't mind the developers answer. In fact that's why it is 
> posted to this
> list.
> 
> This is a completely SIP scenario. From what I have read so 
> far, it seems
> this is possible using a "reinvite" but I have not been able 
> to figure out
> how to force it.
> 
> Has anyone tied to do this before? Any pointers?
> 
> >
> > Caller --> Broadvoice --> Cell Phone
> >
> > Vs.
> >
> > Caller --> Broadvoice --> Asterisk --> Cell Phone

It depends exactly what these diagrams represent.  If these
show the RTP stream (leaving signalling still passing through
Asterisk), then you are right - a reINVITE should allow the
SDP to be renegotiated to bypass Asterisk.

If these diagrams represent both media and signalling, then
you could make the transfer using a REFER.  The exact nature
of the REFER (whether it has a Replaces header or not) would
depend on whether Asterisk had already dialled the Cell phone
or not.

For more details of transfers, I suggest looking at:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-07.txt

The Service Examples covers transfer in sections 2.4 and 2.5.

Michael