[Asterisk-Users] Audio cut off at beginning of call

[snacktime]

On 5/2/05, Robert Goodyear <me at jrob.net> wrote:
> 
> 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?

I have had this same issue also on SIP and IAX calls, but it varies
provider to provider.  Last time I checked I had this issue with
livevoip and teliax, but not with voicepulse.  Which is curious
because you had this with voicepulse right?  Maybe they fixed this
problem and the others just haven't caught on yet?