Ross, On 09/30/2010 10:07 AM, Ross Beer wrote:
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.
The answer to that question depends on what exactly you mean by attended transfer, and what the mechanism used is.
Sometimes it's as simple as the phone or the PBX sending another INVITE, establishing the call, and then sending reinvites on both call legs to bridge media amongst themselves while staying in the signaling path. Or, it might mean a REFER with replaces. Do you know the mechanics?
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