[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?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/ddbfcfc3/attachment.htm
More information about the asterisk-users
mailing list