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.


Why do you have NAT in the first place ? The easiest technique is to avoid NAT in the VoIP environment entirely. If your * server is behind NAT - either use VPN or IP ACLs in the sip.conf. If all of your clients are behind NAT - put the server in their LAN and make him multihomed.

One way audio, RTP debugging, ALG fighting - this is a hardcore that is easier to avoid than to handle.


It looks like all of my settings were correct I experimented by adjusting the qualify frequency from 60 to 1 and setting NAT to yes, and everything seems to be working as anticipated.

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