[Asterisk-Users] Call quality problems

[Doug Lytle]

Michael Welter wrote:
>
>
> I'm not on site, but I remember 1.6.4.
>
I had in place 1.6.2, and had way to many problems with it.  I reverted 
back to 1.5.2 and things cleared up.


> Is the phone (or Asterisk) performing echo suppression that drops the
> last part of the tone?
>
I believe the phone does some E.C. along with Asterisk.

> Also, there are no ZAP cards in the system.  What timing source does SIP
> use to play the incoming media stream?

I think the only time you need a timing source is if you are mixing 
audio streams, i.e. meetme, MOH.  In which case you'd probably need to 
run ztdummy.

Doug

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