[asterisk-users] Is Asterisk ready for Prime-Time?

RE Kushner List Account lists at darl.com
Thu Mar 20 14:56:38 CDT 2008


Steve Totaro wrote:
> On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account <lists at darl.com> wrote:
>   
>> Al Baker wrote:
>>  > Quote"
>>  >
>>  > This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
>>  > have about 1,800 DID numbers pointed at it, "
>>  >
>>  > Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into that box, 1,800  DIDs pointing to it sems like
>>  > one hell of a congestion problem and a Dialplan thicker than War and Peace
>>  >
>>  >
>>  I said DID numbers, they point to a PRI trunk group to a T400P, then the
>>  calls go IAX2 to other boxes for processing based on NPA/NXX.
>>
>>  IE: exten=>_906586XXXX,1,Dial,IAX2/un:pw at asterisk50/${EXTEN}@ninezerosix
>>
>>  And if anything comes in for something not configured this catches it
>>
>>  exten => _NXXXXXXXXX,1,Dial,sip/sipdebug/s
>>
>>  If you figure standard telco usage patters, 92 channels @ 25:1 ratio, I
>>  have quite a bit of headroom.
>>
>>
>>
>>  -Ron
>>
>>     
>
> You don't run into choppy audio with IAX that way?  I see that alot
> and the simple solution is to switch to SIP, almost always solves the
> problem right away.
>   

Not really, but both ends have zaptel hardware.   I'm really surprised 
IAX2 connects and functions to these 1.4 and 1.6 beta servers from a Pre 
1.0 machine.

-Ron




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