Hi, I have a few asterisk servers providing some basic SIP trunking and routing. We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing. I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX. I'm looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways. Please let me know what further information would be helpful. Thanks Kenny Watson

Hi Fred, Thanks for the quick response. I already do use some Kamailio features on our internal network for load balancing. The use case that I'm interested in is to effectively replace an asterisk server that I use for SIP trunking to remote phone systems with a Kamailio registrar/proxy and a bank of asterisk servers placing calls direct to extensions on the remote PBX. I currently have this running on asterisk which I route to the different remote PBX extensions using prefix based routing down to the destination peer on asterisk which is essentially what I need to replicate on Kamailio. i.e. 2021XXXX routes to XXXX@remotepbx1 remotepbx1 maybe defined as either by IP address or via a "normal" registered sip peer with a username/password combo. I understand that I can dial a registered device directly but its how to call a remote extension on a registered device via Kamailio. Thanks Kenny Watson -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Fred Posner Sent: 23 October 2014 16:00 To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX Hi Kenny, This depends on the carriers and scenarios that you may use. I know "depends" is a horrible answer, but one of the great aspects of Kamailio is the flexibility of the modules. Some deployments may have a group of Asterisk servers all configured similarly for handling calls. With this type of scenario, you would benefit from using the dispatcher module. Many people like to use Kamailio on the public side of their network and keep their asterisk servers on the private. This would be an example of when to use rtpproxy (in bridge mode). Some carriers hate seeing the chain of systems on your network (ie the asterisk boxes). Sometimes the use of TOPOH helps to integrate with the carriers who have chosen their own "interpretations" of RFC for "security." And there's more... The bottom line, is that the devil is in the details. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/23/2014 09:12 AM, Kenny Watson wrote: > Hi, > > > I have a few asterisk servers providing some basic SIP trunking and > routing. > > We have remote PBXs trunked onto asterisk which calls come into > asterisk and are routing down to extensions on the remote PBX via > prefix routing. > > I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer > which Invites come into and are routed out to either SIP phones which > are registered or to the remote PBX. > > I'm looking for some advice as to which modules would be best to use > to achieve this as the remote PBXs will be dynamically registered > rather than fixed gateways. > > Please let me know what further information would be helpful. > > Thanks > > Kenny Watson >

What I need to figure out is how to on kamailo maybe using a dial prefix specify that the call is going to a remote extension on a "user" and rewrite the to header to be extension@useripaddress rather than user@useripaddress.