http://www.vocal.com/sip-1/call-transferring/
On Thu, Nov 22, 2012 at 10:58 AM, Dmytro Bogovych <dmytro.bogovych@gmail.com
wrote:
REFER http://tools.ietf.org/html/rfc3515
On Thu, Nov 22, 2012 at 3:55 PM, Grant Bagdasarian GB@cm.nl wrote:
Hello,
I’ve been searching the internet to find an explanation on how SIP
transfer
works using Re-INVITE and/or UPDATE, but I can’t seem to find a good
source.
From what I understand(and this is the way we do it), the following
happens:
Bob=Caller
Alice=Called
John=Transfer party
Bob calls Alice. The usual INVITE,Trying,200 OK, ACK.
Alice transfers the call to John using Re-INVITE.
a. Alice calls John. The usual INVITE,Trying,200 OK, ACK.
b. Alice Re-INVITEs Bob using INVITE with adjusted SDP.
- Bob is connected to John through Alice in some magical way. I’m
guessing because the SDP has been changed and for some reason the RTP
stream
flows between Bob and John through Alice?
Is this correct? If not, perhaps someone could explain it to me from scratch.
Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts).
Thanks,
Grant
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users