[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
More information about the asterisk-users
mailing list