[Asterisk-Users] Limiting simultaneous inbound SIP calls

[Philipp von Klitzing]

Hi!

> Initial thoughts are to use a counter, increment on call presentation,
> decrement on call tear down, and give the inbound call busy or congestion
> treatment if the counter is above a certain value when the call is
> presented?

I guess you need to "protect" your rather thin Internet uplink so that 
"one call too many" doesn't bring all existing calls down? The above 
approach will give you trouble with transferred or prematurely 
disconnected calls I assume...

But you could use AGI and/or a SYSTEM command to do SHOW CHANNELS or 
something similar and that way determine what exactly is going on on your 
box. Another approach would be to use the manager API for this, however 
the first approach looks easier and more advisable to me for your case.

Cheers, Philipp