[SR-Users] Attended transfer in load-balanced multi-server setup

[Pan B. Christensen]

From: Iñaki Baz Castillo

2011/4/13 Pan B. Christensen <pan at ibidium.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).

...
The Asterisk servers named nortel1 and nortel2 are not PBXes, they are 
gateways to the old Nortel PBX system. Their main purpose is to bridge SIP 
and ISDN PRI (qsig). PBX features reside on a different set of Asterisk 
servers, which have already been changed to failover instead of 
load-balanced because of issues with calling queues.
...

It's not possible your scenario. It's certainly hard to balance
traffic when you need a PBX role.

...
I was fearing that answer. The customer wants the whole system 
load-balanced, but I've found this difficult in Asterisk on several 
occations. Perhaps I'll have to convince them to change these servers also 
to a failover design, but that'll require interface cards for an additional 
8 PRI's in both ends.
...

> 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.

...
Thank you very much for your reply.

With kind regards,
Pan B. Christensen