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

[Robert Goodyear]

On May 3, 2005, at 6:32 PM, snacktime wrote:

> 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?
>


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