is it possible to support 25,000 phones and 15,000 concurrent calls?
Not with a B2BUA, look to a true proxy like kamailio
thanks for the tip
At that expected rate, you might want to add clustered proxies (of kamailio) and well set srv dns records to distribute the load, don’t expect much ‘pbx function’ though.
It will support SLT type feature with voicemail?
That’s all I’m looking for.
What’s your thoughts about loading this on AWS
no, for any media support the destination will need to support that, possibly freeswitch when well clustered could do that, but not a b2bua
personally I think AWS sucks, I use Digitalocean or google
You have a big project in place, are you truly prepared?
You are going to need a much more detailed specification, as you are going to have accept some compromises in order to create a scaleable solution. You didn’t even say that this was SIP.
When you say 15,000 calls, these can’t all be connected internal calls, which couldn’t exceed 12,500. If they are mainly external calls, and there is a trunk that you didn’t mention, I would point out that each call will use 4 UDP ports and there are only 65,535 available for all purposed, a significant number of of which are reserved for various purposed, so you would be close to the limit for a single IP address.
True SIP proxies don’t deal with feature codes, although it might be that some software intended for primarily proxy use does have some capability for that. Typically you would have a farm of PABX machines to do the higher level functions, behind proxies which do the load balancing, and probably don’t handle any media or in call signalling, themselves. Voicemail requires something to handle media.
How are you going to get 15,000 concurrent calls? Are these internal calls?
Please describe your application. The only thing I can imagine with 60% utilization would be a large inbound call center, but that wouldn’t involve “SLT features”.
It is impossible for Asterisk to realistically provision that large an infrastructure by any means, End of story 
For sure, anything close to 15,000 concurrent calls is impossible, though if the application could be configured to use direct media (RTP passes directly between extensions and trunks, Asterisk is not involved), you could come close. Of course, this means no recording, transcoding or encryption, and probably means no in-call DTMF features. We’d also need to assume 200+ seconds ACD, so you’d never exceed 75 CPS.
However, suppose you need to design a system to serve the room phones in the five largest Las Vegas hotels (27,548 rooms). You use the existing analog phones, connecting them to ~600 48-port FXS gateways. You configure each gateway as a trunk (one registration), rather than 48 extensions. I would guess that this system would never have more than 300 concurrent calls, which (without transcoding) could be comfortably handled by a large server (though in real life, you’d probably have one per hotel).
Probably not the right answer, but this could scale out it seems like. Although I cannot find anyone taking it as far as you need.
seriously you guys, a b2bua is just not feasible here, asterisk is a b2bua , do I have to say more ?
Okay, okay, but what about FreePBX then? 
Just not the right tool, it can’t scale that big. Kamailio can act as a media server
but this use case would will need to be one big muther^$^&er
I get that this sample config is still on GitHub but the voicemail module has been deprecate and removed from Kamailio for quite some time now. It’s not in 4.x or 5.x, you cannot compile vm.so anymore.
I would think it unwise to have a large, geographically concentrated. group of users, who could use a LAN, reliant on cloud server.
Are all five hotels under the same parent company? Are all five owned by the same owner? I do a lot of hotels and what is required for a Best Western isn’t the same as Hilton or Choice.
I cannot see a reason you would use one PBX for five different hotels. Regardless of size.
I missed that it was five.
The number doesn’t really matter, it could have been two. Even then you are running into collisions because even two hotels will have overlapping rooms which will be an issue for PMS, wake up calls, voicemail or anything room based for the phone system.
Regardless, I am still waiting for how there will be 15,000 concurrent calls happening. At a 80% ASR, that would mean there would need to be about 310ish calls per second with 80% ASR equaling 250ish answer calls per second. That’s what you need to sustain 15K concurrent calls, 250 calls per second being connected.
How is this being achieved and what type of calls are these? PSTN destinations?