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can you help in doing the following.
1 server as asterisk and open vpn server...and multiple linux asterisk boxes at branch locations with open vpn clients.
the asterisk server will route the call towards the branch asterisk machine where the connected voice gateway behind the branch asterisk server will receive the call to terminate.
compression is required between these asterisk to save bandwidth.
configuring the asterisk to make dial plan to receive call and the route it gateways.