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
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety."
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