Building 1st system

I’m trying to get a baseline of what hardware I’ll need to build out a system for 80-100 users with Voicemail. I have Aastra IP phones so I’d just need server and cards I assume, just need some guidance.

Thank you!

It’s really tough to give someone guidance when starting from the beginning. The number of sip devices is irrelevant, you’ll need to base your hardware on expected call volume, queues, applications, etc. Not sure what cards you’re suggesting; pri cards, fxs/fxo cards. You could eliminate the cards and use external equipment like a digium or grandstream gateway. For example, I run 170 sip devices on an intel nuc (4 core i7, 16 gb RAM, and 2 ssd drives with a second nuc as failover). It rarely sees more than 60 simultaneous calls and the cpu never spikes over 20% (of one core). We’re going to have 5000-7000 devices so I’m using vms with 8 cores and 20 gb ram and set a soft limit of 400 devices per vm.

Understood.

Despite doing a fair amount of reading though, I still do not have a clear picture of the hardware I need, but it appears to be at least a 1TE133F inreface card and 1TCE400BLF voice compression.

Does that sound like a good start to support 8 concurrent calls on a super trunk?

Thanks for your response.

I’m not sure that you would need a voice compression card. If you’re only going to have 8 concurrent calls, a half pri would suffice and any transcoding could be done on the server. I believe the codec from the CO will be uLaw (G.711), so if you’re using a different codec for your internal devices, something (server or card) will need to transcode. Or you could just configure your devices to use the same codec as the CO. I’m not trying to talk you out of the card. I’ve never used one. I’m just not sure it’s necessary.

Even better!
So if ail just wanted the 8 concurrent calls, I’d be fine with just the 1TE133F interface card?

Likely so. If you allow your devices to use all of the codecs, it should negotiate the codec to the incoming call anyway. For example, I use G.722 as my primary codec, but I allow G.711 on each device and in the sip config. So when a call comes in from the CO, the server will see that it’s uLaw, and negotiate that to the endpoint. However, when I make a call outbound, it may transcode the G.722 to G.711. I’d have to check (maybe someone else could already confirm), but it may negotiate back to the device to use the same codec. I’ve run 20 concurrent calls on a Raspberry Pi.

As for testing concurrent connections on your hardware, I’d recommend using SIPp.

thanks @mkozusnik i appreciate your input