Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2 incoming calls.
The prompts are stored in ulaw format (and the IAX2 calls use ulaw).
The
On Tue, Nov 24, 2009 at 11:47 AM, Dr. Michael J. Chudobiak
m...@avtechpulse.com wrote:
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2
On 11/24/2009 02:14 PM, David Backeberg wrote:
The asterisk console claims that the IVR prompts are proceeding in the
expected fashion, but I can't hear anything.
Are you playing with the system clock?
...
dramatic ntp changes?
No, that shouldn't be happening. But I'll keep it in mind while
On 11/24/2009 02:14 PM, David Backeberg wrote:
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
Are you playing with the system clock?
Actually, setting the internal_timing option seems to have fixed
On 25/11/09 5:47 AM, Dr. Michael J. Chudobiak wrote:
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2 incoming calls.
The prompts are
Hmm, out of curiosity, does anyone have experience with call recording and 3ware 9550SX cards? We're running a raid1 mirror.Our call recording load is more like 3-5 than 30-50, but I feel that there is some correlation between the two anyway at this point.
Thanks.On 6/12/06, Steve Totaro [EMAIL
Hey All,I've been experiencing a problem for a bit. During a call to the PSTN, audio will cut out for 2-5 seconds. It's completely random and may or may not happen during a call.Our setup is 79XX phones - asterisk - 2811 router - PRI to the PSTN. Everything is talking SIP. The asterisk box is a
Gary,
I would check echo cancelling parameters first. I've seen this to happen
with one of the zaptel echo cancellers. Try to change the default echo
algorithm in zconfig.h, and recompile and install new zaptel. Also
zapata.conf echo parameters may need to be changed either way.
Andrei
We're not using any zaptel hardware though. I didn't think the echo cancellers would be doing anything? We're digital and sip from end to end. Do I need to disable echo cancellation in some way?
Thanks.On 6/12/06, Andrei (MPI) [EMAIL PROTECTED] wrote:
Gary,I would check echo cancelling parameters
Recording many simultaneous calls can cause this behavior too.
Thanks,
Steve Totaro
Gary Richardson wrote:
We're not using any zaptel hardware though. I didn't think the echo
cancellers would be doing anything? We're digital and sip from end to
end. Do I need to disable echo cancellation in
That could be an issue. Would recording to a ram drive solve the problem?Thanks.On 6/12/06, Steve Totaro
[EMAIL PROTECTED] wrote:Recording many simultaneous calls can cause this behavior too.
Thanks,Steve Totaro
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We battled this same issue for a couple weeks, at about 30-50
simultaneous recordings the audio would get all screwy.
I looked at that solution but opted for something a little more
passive. I use orkaudio to sniff rtp streams and mux them. I have it
to perfect quality, the same as the
has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?
The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)
The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its
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