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I have an asterisk server (Elastix on CentOS 7) that I am currently running in Amazon Web Services. The server works great and clients can connect without issue using a stun server. However I have several Digium phones (D-50) that I need to connect. These phones do not have an option to use a stun setting that I have been able to find, and currently when behind our firewall only have one way audio.

I have tried to find a solution but so far have been unsuccessful. Are there any changes I Can make server side or client side to facilitate this?

I could forward ports, but as far as I can tell that would only work for one phone and I need to use several.

Joel Lewis
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1 Answers1

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Actually, we've found that the easiest answer to putting a phone behind NAT to connect to an Asterisk server on a public internet server is to set the SIP registration timeout to less than 120 seconds (usually 100 seconds). This ensures that NAT is always forwarding the port back to the phone. It also ensures that the phone's private IP address is always up to date in the NAT settings, and basically, that everything "just works".

Other schemes like STUN don't work anywhere near as well, and this solution is really quite robust.

Ernie
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