16 Apr
2011
16 Apr
'11
3:20 p.m.
2011/4/13 Pan B. Christensen <pan(a)ibidium.no>no>:
Scenarios:
A call comes in from Nortel/PSTN (user A) through Nortel1 to a SIP device
(user B). User B pushes the transfer button (puts current call on hold and
makes a new call) and dials a Nortel / PSTN number to user C. This call goes
out through the nortel2 server. Users B and C talk a short while before user
B pushes the Transfer button again to connect users A and C. In this
scenario, the REFER is forwarded by kamailio to nortel1, which replies
"SIP/2.0 481 Call leg/transaction does not exist." The call to join is of
course on the other asterisk (nortel2).
The problem here is just in Nortel 1. It receives an in-dialog REFER
through the dialog between user A and B, and such dialog contains a
"replaces" parameter so user B can generate an INVITE with "Replaces"
header to user C and the attended transfer completes.
But for this to work you should use a single Nortel instance rather
than two or more (I assume that the Nortel behaves as a PBX). The
REFER Nortel (User 1) receives indicates it to call to another PSTN
number, so it would generate a new call to the PSTN (rather than
routing such an INVITE to the other Nortel box).
It's not possible your scenario. It's certainly hard to balance
traffic when you need a PBX role.
A similar scenario is where user B transfers to
another SIP user. This call
will only exist in one of the kamailio servers, and Asterisk will give the
same response to the REFER.
A third scenario is where both calls are handled by the same Asterisk
server. This scenario works.
I'm assuming I'll have to build REFER-handling logic into kamailio, but am
unsure of how to proceed. Any suggestions?
No, Kamailio can do nothing here. REFER is just an in-dialog request
Kamailio correctly routes in the loose_route section. The problem is
not the proxy, the problem is the explained above.
Regards.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>