How to configure for internal extension VOIP only?

We only require an internal voip on our local area network, we do not have an Internet connection nor do we want to go outside of our LAN. I have the free pbx web admin up and running, have filled in some of the settings. Using softphone program right now X-Lite, which when we try and call either of the two extensions set up gives back the message: "…number is out of service."
I am using SIP, although after reading documentation, I am wondering if I should use IAX instead.
I have a zap channel running. And when I go into the User GUI interface it does show the calls being made between the extensions under the call monitor…so that indicates to me that the phones are at least getting to the pbx.
To sum up my question: What settings would I use for NAT since I am not using/needing external phone access. I set as Public, with my local area network. And would I use the extension number as the DID in the Outbound settings? or do I need a DID?
Any assistance would be greatly appreciated. I am trying to get the softphone VOIP working before I go on with trying to configure a phone, as we only have CISCO phones and I understand they are difficult with anything but CM. Thanks again.

If you have no internet connection, then don’t worry about NAT, (there is none) ensure that all your devices are on the same network as your server, if they don’t register, then you are doing something wrong, don’t over-complicate, all the extension needs is a username (extension) and an authority (password) that agrees with the server, if you use softphones make sure they are on different IP/ports or it won’t work, if you have a firewall on the machines that host the softphones, allow SIP and RTP traffic through it.

Got back to work today, and I guess yesterday I did change NAT to Never. So still have same problem. Firewalls are off on both of the PC’s I am testing. Extensions with passwords are set on both the PBX admin and the softphone’s. Using X-Lite soft phones. Have an Inbound route pointing to the ZAP Channel Trunk, and Outbound route pointing to same Trunk. I don’t remember actually creating the trunk, it is a ZAP Trunk (DAHDi compatibility Mode). Could it be the wrong kind of trunk? Even though the PBX admin is monitoring the calls from both machines… Thanks again, I know nothing about VOIP.

Maybe you should start by fully digesting the links in

It is helpful for people who know nothing about VOIP.

Thanks for that, I have tried to fully digest the above. Unfortunately the powers that be do not have time for me to fully digest the information required to get this up and running. This would have been much easier if I had the opportunity to plan it, purchase the right equipment, and learn what I am doing before I had to do it. Due to poor planning on their part this has become a crisis for me. Thanks anyway, I will go over to Experts Exchange where they are more willing and perhaps able to assist me.

The paid support option from Schmooze might prove more effective for you in this case

Then the powers that be should empower you to pay for support/consultant.

You have no clue what you are doing and you expect us to do your work for you while you get paid for it.

Why would you need a trunk for your application? More importantly why would you send traffic out the trunk and expect it to come back in?

No Zap0 is a hardware resource BTW. If you explained what you are trying to do you might get help also.

My guess is you voted for Obama

P.S. I have never gotten Experts Exchange, where for points and T-shirts, seemingly knowledgeable people are willing to help folks who won’t help themselves. Sometimes on complex enterprise level stuff.

Not everyone is American so did not vote for Obama. Obviously powers that be do not have the $ nor do we have the time to seek approval from the many layers of power. Need this to allow communication out in the middle of nowhere, I don’t know why they just don’t use the openfire Chat that I already have running on same machines but they want phones (20).
All I am trying to do is get two soft phones (for now) to communicate using the free pbx/asterisk software over a local area network with no outside access. I started with asterisk because it has been used in Afghan in a similar situation. I think I could do this easier and simplier at home with Internet, but we will not have Internet for this as it is a training scenario. True I have no clue of VOIP, but somewhat of a clue of networks, did manage to get the free pbx linux machine running and communicating with the network, also a terrabeam to shoot the network out into the middle of nowhere; not asking anyone to do my job just asking if anyone was in the same situation and what might have worked for them. I believe that Experts Exch does that also, I have assisted others and back when they started - many years ago they did not have t-shirts, points, etc. It would be considered a forum back then, like this.


Then good luck,

You ask to implement the simplest thing, two extensions on FreePBX/Asterisk and two softphones, you have no need of trunks, inbound routes, outbound routes etc.
These basics are covered in the documentation well. For the softphones there is any number of posts out there on making them talk to asterisk/FreePBX try .

Given that and as you have noted that you know nothing about VOIP and don’t have the time to learn, then yes you will need to get someone else to do that thinking for you, that’s why I suggested Schmooze as few will know as much about FreePBX than them.

Well, you don’t have to do anything to allow extensions to call each other.

You say the phones are in the same LAN as the server so they don’t require a gateway to reach it?

Are the phones registered? If so you don’t need routes or anything the extensions should just be able to dial each other.

Do you have the two extensions registered?