How many phones freepbx/asterisk can manage with a single istance?

How many phones freepbx/asterisk can manage with a single istance?

I’m pretty new to asterisk but with freepbx the configurations seems to be very simple.

Now I have only 70 SIP phones here in headquarter, but I’d like (it’s more a dream…) to migrate to voip all traditional phones in branch offices and centrally administer everything.

Is it possible to do the job with only one freepbx at the headquarter? with the reinvite parameters set to yes, I hope that a site to site call doesn’t waste the bandwidth to headquarter

Asterisk and FreePBX can handle 100’s of phones. Asterisk is a B2BUA so if you use reinvite you take Asterisk out of the audio stream and lose many of the features.

This comes down to diminsoning and there is no simple answer. You can put several hundred channels on a single instance of Asterisk with proper hardware but if you have a Raspberry pi expect 3 or 4.

James I like dim som too but in this case I think we are talking about dimensioning.

It’s an art rather than a science. Concurrent calls is much more important than registrations. CODEC complexity, recording and audio mixing (conferencing, DISA, barge/whisper) all take CPU cycles.

The good news is that fundamentally Asterisk was designed to run on a P3 so a good dual core Atom box is remarkably capable. A current generation (is there a Haswell core Xeon yet) server with 15k SAS drives and 8 cores can handles hundreds if not a thousand concurrent medium complexity CODEC calls.

I agree with Scott, I would add to that, with larger loads, you might well need to tune your ulimits also, Asterisk is quite prolific in using file handles and the kernel defaults might well run out after a few hundred concurrent calls especially with transcoding, conferencing, DISA, barge/whisper etc. (unicast paging will often kill you, even on small systems :wink: )

Thank you all for answers. the first attempt was not so exciting :slight_smile:
I tried to make a call from headquarter to Site A and they can’t hear me.
This is a portion of my network, so you can see the routing path.

For now, the only change I did to solve my issue was to add the Site A subnet in Sip settings → Local Networks.

I’m starting to think that it will be more complex than I believed.
In the current configuration of all extensions, the “Can reinvire” parameter is set to NO.
I think that I need to switch that parameter to yes on all extensions to preserve headquarter bandwidth, so internal calls in site A remain in Site A without waste the bandwidth.

But what happen if the call will be originated from Site A to Site B??? There is no route between Sites. Infact all sites are able to ping only Headquarter and the Hot Site.
Do I have to review the entire routing table and enable a full routing between sites? Do I have to create a different internal context for every site?

This is not relevant to your posts subject, maybe start a new thread but read the wiki and search Google first, until your network works passing rtp traffic successfully then reinviting/not reinviting is a moot point.

If you want to use reinvite you will need all the networks to be able to talk to each other.

As I mentioned when you use reinvite you lose most of the useful features that Asterisk provides since Asterisk will no longer be in the media path.

Sorry, it was not in my intention to go out of topic.
I was asking how many phones I can manage with a single freepbx istance, because I’d like to plan a migration to voip.
Now, by your answers, I know that my server is able to manage all the site’s phones.

The next step, for me, is to understand the best way to deploy the solution…

So I’ll open a new thread.

I would suggest that you might better look at a real SIP proxy solution for your use case.

Many thanks. Sip proxy may be a solution in my scenario.
We have pfsense firewall appliance in every site and I found that they are capable to run Siproxd. I’m only concerned about the hardware equipment, because they all are P3 with 512MB of ram.

As alan said, we loose most of asterisk functionality using reinvite, and I think that with this trick(sip proxy) we loose in functionality too, since Asterisk will no longer be in the media path. Is that true?

Actually I was more suggesting something like Kamailio (OpenSer), siproxd is a nice little daemon to handle multiple SIP registrations behind a NAT box. You might be surprised by how a Kamailio or three can help you with your HQ/HotSite/Other Region topology. It makes a very solid failsafe front end to FreePBXi but you will definitely need to RTFM there :slight_smile: Another option with PFSense would be FreeSwitch which is a package away on those boxes.

many many many tanks

Oh dear! are you Russian? :slight_smile:

Post must be at last 20 caracters, so I wrote many many :smiley:
Unfortunately my English is worse than my knowledge of asterisk/freepbx.
I’m the only Italian man that don’t know english :wink: