I don’t know enough this rules on the USA. Sorry.
But if we can do something, I guess.
Anyway, DIDs are on the same office, so for the 911, it should be doable I think.
Anyway, In France we have to send only the CID numbers (Primary and others DIDs) allowed by the trunk.
Before, any PBX could send any CID number. Today, it’s not permit. Just for avoid to present another CID Number e.g: 080000000 instead of 0240404040.
It’s managed on the trunk anyway.
The US Regulations don’t see it that way… Just having a limit a the PBX is not enough to be in compliance.
I wonder how to become a SIP provider. It should not be easy. 
No it shouldn’t. Since you’re in France you would have to look at the rules for France. Just like anyone else would need to look into the rules/requirements in whatever country they are in.
I mean in the US, for the longest time, VoIP providers were left to run wild but that has changed in the last 5-6 years. Your design layout is exactly how many small “providers” did this in the US until the FCC dropped the hammer.
No there is no problem in France because the main SIP provider connected to the FreePBX main applies theses rules. It’s not exactly like you in US.
Got it. It makes sense.
I think the idea of interconnecting privately several PBXes scattered within different facilities (even located in different countries) of one company and privately routing calls and exiting them judiciously to lower cost and improve reliability is perfectly feasible both here in the US, in EU and most countries.
That wouldn’t necessarily require a telephone service provider status.
I’ll leave it to that, but if anyone has concerns regarding the regulatory aspects of a setup like that, again from a user/customer point of view (as opposed to the one of telephony vendor), I suggest that you seek legal opinion.
@franck.danard , you might want to look at https://dsiprouter.org/ which is basically an easy GUI into Kamailio which is a SIP proxy which is what you really want here. You can hang any number of FreePBI or other VOIP servers off it and use any Transport including Tailscale for GNAT/NAT/VPN (or whatever) transparantly between them, The PBI can be internally routed and external calls are proxied cleanly on that server (or relatively easily a cluster)
Definitely would consider DSR. Also, ivozprovider can provide this same functionality (not sure if they have failover). While not a dedicated sip proxy, they have Kamailio baked in and can act as a sip trunk to other PBX using either their “retail” or “wholesale” customer functionality.
One of the really nice open source, multi-tenant asterisk-based PBX I’ve come across
Hey Dicko.
You can use any SBC too. But here, I’m talking about FreePBX module. That’s it. 
Hey Franck, I appreciate that but I find using proxies where possible over B2BUA’s for both robustness and bandwidth, but if media needs to be ubiquitous and yours is, then excellent!!
This module can be usefull anyway. But I know it’s not the best way for certains cases. Let’s say it"s an atlernate way to start a SIP provider.
Also, It can be used simulating a SIP provider.
In what ways will this simulate a SIP provider?
Just imagine for what it could be used.
I see you don’t like this module. Whatever. You don’t like it. But someone else like this module.