I am trying to get a home asterisk server set up with a sip trunk group that i will use for LD. The server is behind a firewall. Calls between UAs in my local network work fine. I am able to set up calls on the external sip trunk group but no sound.
Obviously some sort of NAT issue.
Sniffing the packets on the external side of the router, i can see that the SDP portion in the SIP invite is using the internal 192.168.x.x address and not the external one that i have specified. I suspect this is why i never get RTP back.
in the [general] section of sip.conf i have:
externalip=(my external ip, at least for now)
localnet=192.168.1.0/255.255.255.0
and also
nat=yes (but i think this has more to do with how asterisk responds to invites then how it initiates calls)
shouldnt that be enough?
thanks!!