[asterisk-users] handling SIP trunk with limited concurent calls

[Benoit Plessis]

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