Develop and Configure Kamailio SIP Server

I currently provide an integrated IVR service based on FreePBX. The FreePBX service will soon be migrated to purely Asterisk. The FreePBX server accepts all incoming calls and looks up a route via a database and processes the call via AGI (PHPAGI). This is redundant to the extent where if the one FreePBX server goes down, the NGNs can be routed to a secondary server where calls will work as intended. My requirement is to install a highly available Kamailio setup in front of the FreePBX servers. I have no experience with Kamailio. I require: · Twin Kamailio servers that accepts calls from two SIP Trunks (one with user/pass/registration string authentication, the other with IP authentication.) · If one Kamailio server fails, the other will continue accepting calls. It would be a bonus if more Kamailio servers can be added later to share the load. · All incoming calls to Kamailio to be distributed evenly to the media servers via the dispatcher module and round robin. If one fails, don’t send calls to this server. · Ability to add more media servers later – I can add these via MySQL or via Siremis. · Calls originating from Asterisk to be passed to the Kamailio server and routed out to the required SIP provider. · A copy of any configuration files, incase the servers need to be redeployed in future. I can provide you with · 2 servers with Kamailio installed – although I would prefer to provide you with clean installs of Ubuntu (preferred)/CentOS to ensure everything is built correctly. · The servers can either be NAT’d or on a private network (I would prefer the media servers on a private network at least) · NGN pointing to the Kamailio IP address · Asterisk server to test calls are routing correctly. If you would like any more information, please do not hesitate to ask.