I have the Asterisk PBX on the DMZ. Calling using IAX2 to IAX2 works great, but SIP doesn’t… Everything connects fine, but there is no sound. The SIP user can hear me (If I call using IAX2) but there is no sound from thier end. If I call a SIP user using SIP there is no sound at either end.
It did work while connected to my home network. Calls to and from IAX and SIP worked fine.
do you have Nat=1 in the Peer definition in your Sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.
Thanks for the reply.
I moved the PBX from behind the router, changed NAT=yes to NAT=1.
Looks like it was the router that was the problem. Calls from outside of the LAN using SIP now work, calls from inside the lan on SIP still don’t.
It is an old router, guess its time for a new one.