On May 3, 2005, at 6:32 PM, snacktime wrote:
On 5/2/05, Robert Goodyear <[EMAIL PROTECTED]> 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?
It might be time for me to do another QA session. It's been a while since I did some A/B testing across my providers. FWIW I use Teliax, VP, VoipJet, SimpleTelecom and I have a few minutes to burn off of sixtel if they're still in business.
I'll let you know what I discover.
/rg
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