Try to figure out how (if possible) to set up 4 (or more) server Asterisk. These server* has a trunk-line to every other server*.
These server* acts as servers for client* that needs to call another client*.
Every client* has its own country code. Every client* has 2 trunks to 2 different server* (for failover).
Client*-001 dials 99003123. (99 for entering an ‘external’ call through one of the trunks) (003 specifies which client* to call) (123 is the extention).
This call should be sent to one of the server* that looks at the 003 and decides to which * he should send the call to. If there is a client*-003 that has a trunk to him, the call should be routed to that client*, otherwise the call should be routed to that server* where the Client*-003 has one of its trunk-lines.
(The above example: Client*-001 has got trunk-line to Server1 and Server2. Client*-003 has got trunk-line to Server3 and Server4. Calls from Client*-001 goes through trunk to Server1 (or 2) and Server1 (or 2) routes to Server3 (or 4) who sends the call through trunk to Client-003.)
If setting up the asterisk network this way, then client* don’t need to set up a new trunk when a new client* is entering this asterisk-network. The client* don’t even need to know if there exists any other clients. Only the server* needs to know about the new client* and where its trunk-lines are. If client*-001 tries to call a client*-999 (that doesn’t exist) then he just gets an error that saids something like “Could not connect”.
Estimation is around 100 client* that would need to be able to reach each other. Estimate peaks of round 75 simultanious calls.
There is not going to be any need for external connections (like client*>server*>pstn or client*>server*>otherSIPserver).
All trunks is going to be in VPN-tunnels.
Is this a good idea, or does it create big problems?
Is there better ways of setting up this kind of phone-network?
Does anybody have any suggestion of the configuration-files in the server* so that they route calls in a good way (if possible, could the * decide which trunk-line he should use based on the fastest/best route for best sound quality)?
Any suggestions on client* configs?
I have tried to find out some docs about this, but I haven’t found any. Suggestions on some docs that describes how somebody else has set up something like this?