You have hardware echo canceling *outside* of your T1 card?
No, on the card.
The DAHDI layer has some buffering that can help with jitter, but the
default buffers can only handle 80ms of jitter. You can increase this by
setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms
On 06/19/2012 04:23 AM, Richard Kenner wrote:
You have hardware echo canceling *outside* of your T1 card?
No, on the card.
Then you definitely don't want 'echocancel=no' set, or you'll disable it.
The DAHDI layer has some buffering that can help with jitter, but the
default buffers can
Interestingly, that isn't completely true. If it goes out a SIP trunk
to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300
(where the T1 goes), it has the same problem. This was leading me to
believe that the problem was on the 8300.
Well, that doesn't disprove my
You have hardware echo canceling *outside* of your T1 card?
No, on the card.
Then you definitely don't want 'echocancel=no' set, or you'll disable it.
When I thought that it was echo cancellers fighting each other, that's
exactly what I wanted to do.
--
On 06/17/2012 06:43 AM, Richard Kenner wrote:
Things work fine when he's talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there's clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds
like an echo canceller
I'm having a wierd clipping issue with one employee who's using a phone
over a satellite Internet. He was sold that system specifically for use
with VoIP. Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter.
Things work fine when he's
