I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing.
I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does not. Ideas?
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