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

[Sherwood McGowan]

Gordon Henderson wrote:
> On Fri, 6 Jun 2008, Benoit Plessis wrote:
>
>> 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 ?
>
> Isn't CONGESTION what you want? And if the remote SIP provider returns 
> CONGESTION, then it ought to return it almost instantly too, so 
> "scanning" a list of SIP providers in-turn, before ending up with a 
> PSTN interface ought to take fractions of a second..
>
> Just don't confuse CONGESTION with BUSY.
>
>> 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
>
> See above - the SIP channels ought to return CONGESTION immediately if 
> they're "full".. (I can't think what else they might return though?)
>
>> i think i should look after another sip provider
>
> I currently use this in 2 applications - one is to a SIP -> GSM box 
> with 2 ports, when each port is busy with a call, it returns 
> CONGESTION, so I try port 1, then port 2, then fall-back to PSTN, (and 
> I had to tell the box to give me CONGESTION in this case rather than 
> BUSY!), and in another application where I do it the other way round - 
> I dial out via 3 analogue lines, but when they're full, Zap/G1 returns 
> CONGESTION and I then dial out via the Internet and a VoIP service.
>
> Gordon
> ------------------------------------------------------------------------
>
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He's right, you should get congestion in less than a second (unless your 
provider is slow anyway in which case you should switch providers anyway).

-- 
Sherwood McGowan
VoIP / Telecom Solutions
sherwood.mcgowan at gmail.com