Freepbx, SIP and NAT

I’ve run freepbx for a while on my home phone system but I’ve never tried to configure an external SIP Trunk.

I remember looking at SIP and NAT and Firewall and never being able to get that going.

Time to try again.

First of all… if I have a Firewall ( linux box doing ip tables nat masquerade ) any my freepbx box is behind it on a 10.x.x.x. network… is it possible to connect to a SIP provider like Flowroute and have two way communications? I saw things like STUN, ICE, and a few other protocols.

I followed the flowroute freepbx SIP tutorial and got a SIP connection going, but calls were not going over the Flowroute Sip Trunk I setup.

I think that SIP works better if you don’t have to NAT and route ports… but will it work good enough so I don’t have to put my freepbx with a public ip address ( I only have 1… so if I do this… the freepbx box has to also be my Internet Router for my home network )

Thanks - jack

Thought I’d get some sort of response…

Is this too much of a noob question?

  • jack

Yes. In my experience, I would say the majority of PBXs are behind a NAT router and obviously they can make calls.

  • Enable NAT, set the external IP address and identify local networks in Asterisk SIP Settings
  • disable any/all SIP ALG in the router.
  • For extensions that are not local to the PBX, you want to enable NAT on the Advanced tab

If calls progress, but you still don’t have audio, enable port forwarding on the router for the SIP signalling port(s) and the full RTP range.

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