I've been having audio breakup problems (on my end) in my Asterisk
tests. I'm not sure of the most likely source of this quality problem.
99% of my LD calls are calling into a tele-conference service called
freeconference.com for group meetings. Its a free phone conference
system that works quite well with pstn phones. I've been using it for
quite some time. But the audio problems after setting up Asterisk and
VoipJet are now unacceptable.
About 30 minutes into the call I have difficulty understanding what
others are saying because their voices are breaking up. Short calls
seem to work fine, quality is good.
I'm using Asterisk 1.0.3 on Whitebox Linux (still a novice)
I'm using:
- Analog phone with Sipura SPA-1001 adapter
- VoipJet for termination.
- ulaw codec
- Comcast cable modem connection
Can someone help me narrow down where to begin to solve this? Should I
try another termination provider? SPA-1001 settings? Different
Asterisk settings? Should I provide my Asterisk conf files here? Is a
teleconference service like freeconference.com more likely to have
this type of issue than standard phone calls? Thanks.
--
Lee
