Mike Hammett wrote:
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.
That would be expected since you did not forward the ports used for RTP.
See /etc/asterisk/rtp.conf A sample is in the Asterisk source.
Did you also set localnet= and externip= options in sip.conf [general].
SIP works just fine with NAT if you have it correctly configured and
your server is on a static IP address.
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