[asterisk-users] SIP & NAT

Mike Hammett asterisk-users at ics-il.net
Thu Mar 29 08:51:40 MST 2007


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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