Public Asterisk with NAT Phone Client


#1

I setup my Asterisk server within my LAN and had everything working great, but I could not connect my Phone Client from Remote locations.

Since that was part of what I was trying to do, I moved my Asterisk server to a Public IP address (so it is no longer NATted…) and have all my Phone extensiont (clients) behind my NAT firewall. (That way, no matter where I am, I should be able to connect to my asterisk server the same way.)

I can log in to the server just fine (the X-Lite phone client shows the connection…) Phones ring, and I can answer them. I can speak and be heard by other people.

However, I get no return voice. I can’t hear anything.

I have been looking for the answer, and it seems to be STUN. But I can’t seem to get that working, either.

Does anyone have any ideas as to where I should look for answers? Anyone done anything like this?


#2

You don’t say, but i assume you’re talking about SIP connections.

Have you got two-way voice between your remote phone client and your asterisk server (echo test?)? Have you got two-way voice between the local phone clients and the asterisk server?

If the answer to both these is yes, then you maybe you haven’t got “canreinvite=no” in sip.conf, for all clients involved. I don’t think you’ll ever get the type of setup you’re talking about to work if “canreinvite” is set to “yes”.


#3

I tried to achieve this also and was never able to, at least not with softphones. I had my most success with remote software clients using the IAX protocol, Firefly is currently the only one i know of which supports this.

I got two way voice working with this, but it killed the computer which was running Firefly, at up all the resources even after the call terminated. I had to do a hard reboot as it was frozen.

I havn’t tried again since mainly because I never needed this anyway, but I think STUN is the way to go as you said.


#4

If you havn’t already, have a read of this:

voip-info.org/wiki-NAT+and+VOIP

Pretty much explains it all including solutions/work arounds…


#5

From reading that document it looks like VOIP through any sort of firewall is nearly impossible and probably not worth the trouble.


#6

Well that’s definitely not the case. To a large extent it depends how much control you’ve got over the firewall configuration. If you’ve got no control, it depends on what type of firewall it is.


#7

[quote=“WillKemp”]
Well that’s definitely not the case. To a large extent it depends how much control you’ve got over the firewall configuration. If you’ve got no control, it depends on what type of firewall it is.[/quote]

Very true. You need end to end control. In my case, I had full control where the asterisk server was, but not where I was running the client.

I knew this was the case.

Basically, its not yet at a level where you could just go to anyones network with your laptop and make calls using your * server without tinkering…but its close:).


#8

asteriskguru.com/tutorials/s … erisk.html

Have a look here, it describes different situations and their configurations.


#9

Sonicwall firewalls have excellent VOIP protocol support, especially h.323 and SIP routing/NATing.