I have an Elastix server running in Amazon Web services, but I can't seem to resolve an issue that is causing either one way audio or no inbound calls to ring.
If I set my SIP extensions to "Nat = Yes" they can dial out and audio works fine both ways, however if no phones will ring on inbound calls.
If I set my SIP extensions to "Nat = route - NAT no rport" my phones will ring on inbound calls, but I will have no audio going out to the caller.
I am using Digium D40 and D50 phones, as well as the Bria soft-phone, and unfortunately the Digium phones do not support setting a STUN server.
I have been digging through this guide (http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html) on troubleshooting NAT issues, but nothing seems to work.