Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a "call-limit=2" in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
--
Benoit
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