We are trying to use SER as a NAT traversal solution in front of our Asterisk PSTN gateways and our Asterisk Feature Server. Currently, the feature server is doing all of the dialplan processing. Because of this we can have user configurable dial times, unavailability numbers, forward on busy/unavailable numbers, auto-attendants, 7-digit dialing, etc.
Of course, my life would be simple if Asterisk acted more like a SIP proxy server and the dialplan could be executed somewhat like an SER routing script. Unfortunately, any call that comes from a UA to asterisk creates one call, then asterisk creates another call to the other end of the conversation (other SIP device, PSTN number, whatever), then issues reinvites to the two ends of the conversation to tell them to send RTP to each other.
So what I need, is an SER script that will intelligently handle NAT traversal for UA to UA calls, UA to PSTN calls, and PSTN to UA calls (all going through the Asterisk Dialplan so we can keep all of the end user features) including the damn re-invites. Anybody had to do this before? We are quite willing to pay you for your knowledge if you want to do some design/consulting/coding work for a solution to our problem.
Thanks.