FreePBX in large environment,


I’m wondering if anyone has used FreePBX in larger environment to comment on how well it acted. I’m working on a clients request that requires 100 IP phones and 250 analog phones and was wondering if FreePBX can handle such request.


There are FreePBX installs with a much greater number of phones. No problem at all, provided your hardware is up for it.

In my experience, the analog phones are going to be the long pole in your implementation tent. Having said that, I’ve sold a couple of systems that had the capacity for that and never had a problem.

You might find that partitioning your analog phones into smaller groups and putting them onto individual server might serve you better, especially if you are going to be recording calls.

Depending on your hardware , then you should have no trouble, just google “asterisk dimensioning” for stuff that goes back to the year dot, even in year dot a pentium 4 would handle your needs (it did for me back then for 4 PRI’s on Digium hardware and several SIP trunks and local endpoints).

Thank you all for your responses, I will be using a higher end Dell server with Xeon processor and 16GB ECC RAM.

cynjut, not sure I understand what you mean by “putting them onto individual server”.

Also I was wondering what would be a good choice for ATA’s in those kind of larger environment, I was thinking to use five 48 ports GrandStream ATAs.

For channel banks I would search for adit 600’s on ebay, Dell used to sell them, they are cheap as sh*t and are extraordinarily configurable and the FXS cards are at bargain basement prices because Dell/m$oft no longer support them (adit do :wink: ), the REALLY useful, FXO cards are hard to find but work as well as anything for a couple of hundred bucks or less for each 8.

They are 19 inch rack-able but the ears and connective hardware (two Adits x6 FXS’s for 96 channels in 2U) are hard to find, so use shelves. So you would need 6xAdits and thusly 6U rack space bujt you get some spare pairs, each one has two RJX-25 for legacy 66 block wiring. and a 48v DC power option

(I know all this because I have been using them for years. Not so much anymore, so I have/am trying to, sell them to the cognoscenti)

could be much more phones if the server is powerful enough. you need to estimate max. concurrent calls for hardware determination.

It depends from the usage that you are planning to do. I would suggest to offload the system by using sip to fxs adapter for your analog extensions. Also easier to troubleshoot because fxs ports get “burned” easily so using an adapter would not need to power your system down.

If you keep you system communicating with sip only then with a xeon machine you can handle all your needs.
If you start using dahdi then you have to test first.

Dont forget to to consider bandwidth in your design, depending on that you will also want to consider g729.
We have a couple of systems with 200 extensions, FreePBX handles it well.

Asterisk is perfect for what you want to do.
Someone correct me if I’m mistaken, but in my experience server requirements isn’t really dependent on the number of extensions, its more the number of simultaneous calls you need to handle. And even then if you keep with ulaw the CPU doesn’t do much then either. I know that CPU can increase with call recording sometimes, and conference bridges.
You’ll probably spend far more time fiddling with the FXO adapter than the PBX itself.

Thank you all for your inputs.

agree!, one call can eat about 200KB/s bandwidth by g729

As Dicko says, the Adit 600 is a good way to go for multi channel analog, especially if low cost is important. These units are very reliable, and secondhand ones can be found cheaply enough so that I was able to buy a complete loaded spare chassis for a backup. Currently use a single fxs/ fxo 48-channel unit but in the past have had 96 channels working with a Quad Span TE405 card, (everything sourced secondhand via ebay, ymmv) But the configuration needs to be set manually, and information is very hard to come by.

Don’t know about Adits, but we have two installs that should prove what you are trying to do:

  1. Hotel here in Santa Fe - 90 rooms all being fed Analog hanging off 4 Adtran 924e’s - each Adtran supports 24 Analog ports, so that works out to 96 ports - they only have 90 rooms, so 2 others are FAXes - works like a champ!

  2. Lawfirm - started out here in Albuquerque with 27 phones in 2007 - they now have 460 phones in 38 different states - works like a champ.

Both of these are running on Asterisk - no problem!

Adtran are also an excellent choice, be careful you don’t get an older one because there will be no “far end disconnect supervision” but the Adit’s can do SIP/MGCP as well as TDM (hehe , bad math you can have 6 faxes if you want)

One important thing in large deployments is to set the registration minimum expiry for SIP endpoints to something quite large (min 600) and the registration default expiry to double that (so 1200).
The default setting of 60 (seconds!) means every extension can theoretically refresh the registration at 75% of the default expiry, so every 45 seconds.

Easy to flood the system.

Parameters can be set in

Settings | Asterisk SIP Settings | Chan SIP Settings | Registration settings

Learnt the hard way on a hotel site where the IT guy thought he could plonk a FreePBX in and forget it. Cue network flooding.