I get this problem when I dial out over my voice T1 but not when I dial out over a POTS line. So it looks like in my case it is the voice T1 provider.
----- Original Message ----- From: "Robert Goodyear" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, May 02, 2005 11:52 AM Subject: Re: [Asterisk-Users] Audio cut off at beginning of call > > 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? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users