FreePBX over wan to remote office

First time setting up freePBX. Is it possible to set up a server at the office with no outside calling and have one internal extension [and] have an extension at home , over the wan, and have the extensions call each other with no trunk provider. i just want the extensions to call each other over the wan, no vpn. i understand that the firewall needs to be setup on the near side, but what about at the remote side. are there any special requirements or constraints that will stop this from working. i plan to have a dyndns fqdn for the public ip of the server and enter the username and password in the remote phone. no firewall. thank you

If there are only two extensions and no trunks involved, you really don’t need asterisk, the phones can just dial each other directly.

Would you mind elaborating… My plan is to use a couple of IP phones to dial an extension to the other phone over the wan. How would I dial that phone without a pbx server. very intrigued.
thank you.

Depends on the phone and how they describe it, often it’s called direct ip dialing, or url dialing


You will still need masquerade and forward rules both ends if Nat is involved (just like asterisk needs) , DO NOT use 5060 as your SIP negotiation port or suffer the consequences :wink:

Several providers offer free calling within their network, even on an unfunded account. For example, get an ‘IP Freedom’ account at
Set up two extensions, register your phones to them and you can call between them.

1 Like

Again, the OP just doesn’t need a trunk with a provider or asterisk it’s a waste of money

SIP is totally complete to provide endpoint to endpoint connectivity all on its own.

For people that know what they are doing, yes.

That is not someone who comes in here and posts not even understanding how anything works at all. also. Create two sub accounts. Give the Extensions when created, and then register each phone to a different sub account and you will be able to extension dial between them.

There is the rub :slight_smile: , the OP can learn, most every sipphone or ata comes with documentation of exactly how to do direct ip dialing, if he won’t rtfm then he getsto pay someone else to effectively do it for him unnecessarily.

Depends on his phones. Some can’t emulate being on a public IP, when behind a NAT. Others can’t call without registration. And he would still need a third party provider for dynamic DNS.

Everything I have worked with can, easily and without sweat, as I said natting is the same as asterisk. If needed. Forward your sip port to the extension. (DON’T EVER USE 5060) (Both ends if needed) The rtp will almost always be associated and appropriately forwarded.

(Spa, obihai,. Grandstream, yealink aastra. . ., They all work as advertised, none expect registration please prove me wrong . . .)

And obviously you would need an endpoint identifiable over the internet, that is kinda a no brainier with any number of dynamic up providers.

All of these considerations are identical to how any. Sip transport, including asterisk are necessary

I find freePBX interesting to learn and don’t mind using it. I did not know that you could effectively call between two phones without a phone server. What I wanted to do was have a phone at the office and then a phone I can take with me as I travel around, plug it in to the internet and be able to make/get extension to extension calls from the office no matter where I am. preferably without having to worry about the firewall at the remote end. (because I am traveling). The other twist is that I have an associate that does not have reliable cell service, so I was thinking of connecting him with an ATA and a landline phone. Dont know if that changes anything… Thank you all for the input.

My older Yealink phones, e.g. T26P, do not have an option to not register, nor an option to permit calling when registration has failed. I don’t know whether this changed in newer units.

Some phones such as Cisco and Polycom are not capable of substituting their public IP address into SIP headers and SDP.

In addition, the peer-to-peer scheme requires NATs at both ends that do not modify the source port numbers for the RTP range. This is not possible in a traveling situation, where you may be behind a carrier NAT (mobile data), or a NAT over which you have no administrative control (hotel, etc.).

Also, without registration, the device can’t learn its public IP from a Via Received header, so it would have to use a STUN server (in addition to dynamic DNS).

Yes, Obihai devices don’t have any of these issues and can also do symmetric RTP to work around imperfect NAT traversal at the other end, but the OP specifically wanted IP phones (not ATAs) and IMO Obihai phones are not a good value.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.