Explanation of scenario:
1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B
2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards
3. Number of Server B can be unlimited
4. Number of Gateways/E1 cards per server B can be unlimited
5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider)
6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc.
1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1
2. we need some kind of bandwidth compression system( upto 60-80% than usual SIP calls in g723.1 codec), from Server A to Server B.
3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination
Pls let us know for further information necessery to get the project done. This project actually similar to a provider http://syncswitch.com offering and we we want to provide the same kinda services to our termination provider or tht meant named Server B in our given specifications above. suggest us required Server configurations n OS for Server A to start the projects