Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
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
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit
Another problem i foresee is long delay in dialing sequence when
asterisk will have to dial using 4/5 account
before having a working channel
i think i should look after another sip provider
--
Benoit
begin:vcard
fn:Benoit Plessis
n:Plessis;Benoit
email;internet:[EMAIL PROTECTED]
tel;home:+33 9 52 49 25 06
tel;cell:+33 6 77 42 78 32
x-mozilla-html:FALSE
version:2.1
end:vcard
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
