My two cents worth.
All successful telephone endeavors need to supply “five nines” service , that means that unless you are out of service for less than a couple of minutes a year 99.999% available, your customers will be quite reasonably disgruntle. This is Verizon, AT&T, Bell, or the other PTT’s around the world.
There are no perfect solutions, and certainly no prepackaged open source one, FreePBX/aterisk is as it says a PBX (B2BUA, or Back-toback user agent), I suggest you need a SIP proxy to be behind your various PBX’s (Clients). Kamailio, (an offshoot of OpenSer) is coming close, FreeSwitch is a possible hybrid, your own ss7 switch on the “exchange” closer to reality 
They already have all the hooks to do your multi-tenancy ordering, cdr’s etc. I say hooks, not portals, those you will have to build yourself. All billing will need to be done from this gateway “system” simply becaue it is your gateway device.
Asterisk/FreePBX’s are feature laden but IMHO will never be able to scale to true Multi-Tenency successfuly, Centrex was the Telco’s answer in the 80/90’s but the compromise beteween features and manageabilty lead to it’s almost total demise.
Your SIP proxy should be designed to do the trunking and registrations , the failover and the routing, both IP wise (bgp) and VOIP wise. It also needs to be failsafe in hardware and IP address, although quite trivail they don’t come with a GUI in general yet though ;-).
When you have that working, then feed your children ( function rich PBX’s) from there, All they need to do is register the trunking against your SIP proxy, If they attempt to handle registrations from off network then all the above problems will reappear as your hardware/IP’s appear and disappear, so you need to have the extension endpoints register against your proxy and have that system do the provisioning also.
If you rely on Asterisk, that is a major single point of failure, add HA/corosync/drbd and you have a relatively solid two point failover, major upstream failure of IP routing or VSP failure is largely still out of your control.
The DID’s (origination) are your major revenue stream, make sure they exist on a solid carrier that can handle failover to your available SIP proxy system. (again both IP wise (bgp) and VOIP wise.)
You will have to own that “example- Business Extension with 500 minutes” responsibility ultimately, the small print of every VSP’s “terms of service agreemnet” I know will preclude you from actually using them in a high volume deployment.
You will need multiple failsafe Outgoing (termination) services first for network (IP) then for VOIP. (once again both IP wise (bgp) and VOIP wise.)
Avaya or Mitel et all will bite you for twice that for half those endpoint, they will give you “five nines” though.
Good luck though, It is quite easy to make “four nines” with a cluster of machines doing a SIP Proxy and a bunch of PBX’s, small ones will now happily work on VirtualBox, VMWare or XEN hardware. but not without a lot of knowledge and effort,
If you offer your services at that level on an underdesigned system, perhaps sell your services at half the “big player” going rates, then expect long hours of profitable boredom followed by short periods of intense panic followed by lots of rebates to your clients as you miss your SLA’s
Don’t get me wrong, It is possible, just not as easy as perhaps you imagine 
p.s.
Sorry but with less than 30K I honestly think you expose yourself to a world of hurt as you attemptto manage more than a couple of k endpoints on an Asterisk/FreePBX based system.
regards
dicko