[SR-Users] Asterisk Load balancing and Session Aware

[Ross Beer]

Hi,

I would like to know if the following is possible in Kamailio, I've
tried with OpenSIPs but I don't think it is ideal for my needs.

I would like to load balance multiple asterisk boxes which terminate
and originate calls. To transfer calls by attended transfer any new
calls originating from a phone need to be sent to the same server as
the held call. With the dialogue module I can add the call originating
from asterisk to a profile and the new call from the phone can check
if the user already belongs to a profile and then send the call to the
same gateway.

What I would like to know is if there is a better way to do this or if
Kamailio can perform the transfer without the need to send the call
back to the same asterisk box. I've noticed that the Kamailio dialogue
module has a few more features than its OpenSips counterpart.

I would be very grateful for any feedback.

Regards,

Ross