FreePBX/Asterisk server sizing - Need help

Hello all,

I am trying to design a FreePBX system for about 400 extensions. A large proportion of those extensions will be analog, via 32port fxs gateway devices (13 of them to be exact), with just a few SIP phones. I cannot imagine them to have more than 100-150 concurrent calls.

My question is, what kind of server should I take into consideration?

I was thinking about Xeon 16core processor and 32GB of RAM. It’s important to note, that it will be a BASIC fixed telephony system - no bells and whistles like even call recording.

Thank you all.

Tell us more about how you intend to connect these. That should help us understand what you’re doing with the tech.

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I would start here, but chances are you can do the same with slightly less than what is listed.

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As long as all those fxs ports are normal voice, then @comtech’s suggestion is good. The appliance targeted to your specs would be an i5/8GB (and that probably assumes at least some recording).

16 core/32gb would be over spec’d unless your planning for LOTS of growth.

If the intent is for the box to host other services, plan on using containers/VMs. FreePBX assumes it has control and doesn’t always play well with others.

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Before we get down to hardware, let’s start with this statement. You shouldn’t be imaging, you should know. Is this a new business that is installing a PBX for the first time? Is this an existing business that is migrating their PBX? Is this a business or being used for other purposes? Finally, what is the actual purpose?

I would just like to know what metrics where used to determine the amount of concurrent calls needed. Even if this was some deployment where there would be internal calling from user to user, estimating 100-150 concurrent calls would mean there would have to be high capacity trunking services to support that.


Thanks for the reply.

I intend to connect each of the 32port FXS gateway devices directly to an AT-GS950/24 switch and the customer’s LAN network via an SFP module in the same switch.

All of the gateways will be in a single rack, close to the original PABX and the customer’s main telephone wiring.

Also the FreePBX server will be connected to the AT-GS950/24 and located in the same rack enclosure.

Fair point. However, it is impossible for me to know.

The current phone system the customer uses is GTD1000. Believe it or not. It has no displays anywhere on the system and no way to connect a modern day computer to it. Literally, no IO ports, that I know of. I had a rought ime, finding a picture of it online, let alone some tech info.

It has around 500 analog extensions and God knows how many COs. The customer says they only pay for about 40 PSTNs, so that is my point of reference when it comes to trunks.

Of course, the plan is to move to SIP trunks, but I really have no choice, but for analog extensions to remain analog. Their LAN network is decent, at best, so there really isn’t possible for it to endure a high number of SIP phones.

The company, or the factory rather, is pretty big and they use their PBX system for internal calls, mostly.

Thanks for the reply. The intent is to have a stable system. Nothing less and nothing more. This will be, by far, the biggest system I have ever deployed and I would like it to go as smooth as possible.

No, there will not be any other services. A clean install of debian and FreePBX.

This tells me you’re probably going to be connecting each gateway through a Trunk. We’ve had lots of posts talking about 2x or 4x gateway devices to connect a couple of POTS lines.

I assume you’re doing this to interface to your POTS phones, so setting this up is just going to be a PITA. Once set up, it’s going to be relatively easy to manage (I suspect).

As you move forward, you’ll want to replace the phone instruments with a SIP (or SCCP, if you’re brave :slight_smile: ) phone. When you do that, replace the phones on a “per bank” approach so you can get the old phones out of the network. When you add the new “networked” phone instruments, you will connect them to your PBX as direct extensions.

Not sure if I understand completely. Sorry, english is my second language.

The setup will be something like this:

Provider SIP Trunks -->FreePBX server eth0

FreePBX server eth1 --> AT-GS950/24 <-- each of the FXS gtw devices (there will be 13 of them)


AT-GS950/24 <— SFP port---- customer LAN

Now, the 13x32FSX ports will be wired directly to customer copper telephone installation.

You are absolutely right. It is going to be a “PITA” but hopefully, many many man hours later, it’ll be all set up and like you said, fairly easy to manage.

That’s a regular 10/100 Ethernet network. The 13 FXS Gateways will physically plug into this, right? What are the FXS Devices? How to they logically connect to the PBX? Are they SIP Devices? Are the T1/E1 devices? If SIP, what kind of Ethernet do they use?

I assume there’s a place to plug the rest of the computers in the installation in someplace, which is implied by connecting the SFP ports. That means that the phones FXS Gateways will be on the same network as the computers. That’s fine, but I would go with a physical phone network, plugged into the PBX through a dedicated Ethernet port (eth1) and plugging the “utility” network into a different port (eth2 or eth0, depending on how your provider network is connected). In other words, add another network interface and you don’t need to plug the switch with the FXS Gateways into the main network.

Now, if you plug the PBX into the utility network, you can just plug the AT-GS950 that has all of the gateways into the rest of the network and just subnet from there. Each gateway will have a single IP address, so you’re only going to need 13 addresses to connect your phones (assuming the mysterious FXS gateway that you won’t tell us about uses SIP).

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