On May 1, 2005, at 11:39 AM, Gene Naden wrote:
When we call out from our Asterisk system we consistenly lose the first
roughly 1500 milliseconds of the audio from the destination. This is easiest
to demonstrate with a recorded announcement. In other words, "Hello" for
example is missing.
We are calling over the PSTN via a voice T1 line.
We are using the "stable" cvs from about April 1.
I searched lists.digium.com but did not find anyone with this problem
using the PSTN. Does anyone have any ideas?
Same here, via VoIP. I reported it to the list a while back:
http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088514.html
If you're getting it via ZAP and I'm getting it via VoIP, sorta starting to sound like a setup issue on the Asterisk side, doesn't it?
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