[Asterisk-Users] Connecting 20+ asterisk servers together

[Alex Barnes]

> -----Original Message-----
> From: Vikram Rangnekar [mailto:vicky at freebsdcluster.net] 
> Sent: 09 May 2005 11:26
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Connecting 20+ asterisk servers together
> 
> 
> I have 20+ asterisk servers and need to network them together 
> so a phone on any of the servers can call a phone on any 
> other server without any trouble.
> 
> I can think of IAX trunks between every server. So every 
> server will have an IAX trunk to every server and then prefix 
> bases routing in the dialplan for each server (I can give a 
> number to each server and use that as a prefix for that 
> server). But I think this is a maintainance nightmare and 
> also a very bad approch does anyone have any better ideas, 
> Also should the phones be able to send rtp between each other 
> or only through the Asterisk server since if its through the 
> asterisk server and say an IAX trunk then the max number of 
> calls can be controlled right. 
> 
> Can dundi or the switch statement help me get out of this mess ?


Am I right in saying this is for remote sites rather than 20 servers for
load 
balancing reasons?

As I am about to start hooking up a small number (< 5) sites together
with Asterisk 
servers at each site and am not entirely sure of the best approach.

I was thinking of having extensions for each site something like:

1xxx = site 1
2xxx = site 2
 
then for example server1 would have:

switch => IAX2/user:pass at server2/context

Matching on 2xxx.

But this doesn't sound particularly elegant specially once you start
trying to scale it.
If you do get any other ideas I would be interested to know so that I
can start this 
structure out properly.


Again if this is remote sites, how are the phones going to talk directly
to each other, VPN?
Passing the RTP data over VPN direct to the phones will mean you don't
get the 
benefits of the IAX trunking to reduce bandwidth which would be a shame.
I would be interested to know how people find VPN's for passing audio,
specially if IPSec etc is being used.  I would imagine the quality is
fairly bad on anything except the fattest of pipes.

Is worth noting call monitoring will force the RTP to the servers
regardless, I don't know
if that's an issue for you?


Sorry not a lot of answers there just questions :-D

Cheers

alex


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