OpenSIPS + Freeswitch Class 5 SIP Platform

We require someone experienced in VOIP, specifically OpenSIPS, MediaProxy and Freeswitch to aid in the implementation of a class 5 switch solution. Proposed design; * Two OpenSIPS proxies - All customer termination and origination from / to customer via proxies * Two MediaProxy media servers - All customer RTP is between MediaProxy and Freeswitch * Two Freeswitch SBC's - Acting as PSTN gateways between OpenSIPS and carriers / TDM gateways * OpenSIPS Location / Redirect Server * FreeRadius Server - AAA Proposed design attached as a PDF. Requirements; * Customers can either Register (Dynamic) or trunk directly (Static - IP Auth) with Proxies * Customer Location / AOR needs to be maintained by Location / Redirect server. When call received from carrier by SBC, the SBC must attempt to terminate call to redirect server, redirect server will direct SBC to correct Proxy using 301. * Prepaid / Credit Limits needs to be implemented using CDRTool, Call Control * All calls (Including between customers) must support transcoding via SBCs * LCR needs to be implemented, with all routing information stored within OpenSIPS database - multiple Sofia profiles will exist on the SBC's - Freeswitch should not make routing decisions. * Basic documentation detailing the implementation upon completion Design Considerations; * All routing information, including DID/DDI's needs to be stored in a MySQL database, preferably all within Radius * It is preferable to support call forward (Immediate / Busy / Unavailable) on individual DID/DDI's via AVP's within a database * It would be nice to have geographically distributed MediaProxies in the future - being able to specify a "local" pool of media servers for a customer to use * Multiple number formats will be made available to customers, this will be preconfigured for each customer. When a customer is terminating a call, OpenSIPs should translate and normalize all numbers to E164. When terminating a call to a customer, all numbers must be translated back to the customer preference. How this is to be achieved has not been designed. * On failed calls, customers may elect to either receive only SIP signalling (Ie. 404, 403, 500) with the appropriate cause code or they may elect to receive a recorded announcement instead. How this is to be achieved has not been designed. * There is no requirement to support call transfers, hold, service codes, voicemail or conferencing. This platform is to provide enterprise and wholesale SIP trunking, no Centrex or Hosted PBX services. * There is no requirement to configure IP Failover / HA between servers. It is expected that redundancy will be handled within the call routing itself, it is acceptable that should a proxy fail, customers who are registered to that server are unreachable. Experience Required; * OpenSIPS * AG-Projects MediaProxy, Call Control, CDRTool * Freeswitch * FreeRadius CentOS based Linux servers will be made available via SSH. MySQL cluster will be made available. Network, DNS configuration all done. You will need to install / compile OpenSIPS, MediaProxy and Radius (With MySQL backend). Whoever the project is awarded should be available for ad hoc hourly work after implementation for support / minor changes. An hourly rate for any subsequent work is to be negotiated later.