Hardware echo usually helps. You can aslo try using OSLEC.
- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Friday, October 09, 2009 23:50
Subject: Re: [asterisk-users] choppy sound
Hi,
I am using CentOS
Hi
After a day of running asterisk, I got choppy sound when fw ip-pstn
When I restart asterisk the sound is fine,
Anyone had same problem?
Thanks.
___
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AstriCon 2009 -
...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 09, 2009 3:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] choppy sound
Hi
After a day of running asterisk, I got choppy sound when fw ip-pstn
When I restart asterisk the sound is fine
Nicholas
Sent: Friday, October 09, 2009 11:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] choppy sound
It would be helpful to know the OS, release of Asterisk, hardware, etc.
In my case, I start getting excessive echoes at end of day, so I do
: [asterisk-users] choppy sound
It would be helpful to know the OS, release of Asterisk, hardware, etc.
In my case, I start getting excessive echoes at end of day, so I do a
restart when convenient each morning around 4:00 AM.
_
From: asterisk-users-boun...@lists.digium.com
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository). As a clients I use XLite on Mac, all on the same LAN.
Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM
and plenty of disk space on LEVEL 5 RAID.
Calls to another SIP server (also asterisk)
On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote:
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository). As a clients I use XLite on Mac, all on the same LAN.
Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM
and plenty of disk space on
I had the same problem doing SIP - IAX, 1.4.19.1 as well as the last 1.4.22
In my case I was trying to do FAX and the blips were breaking lots of
the faxes. My solution was to switch to T.38 over SIP and (cross my
fingers) the problems haven't came back so far.
I don't know the source of the
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I don't
have choppy sound
Muiz Motani wrote:
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I
Benoit Panizzon wrote:
Hi there
I europe alaw is usual. I have a SIP Phone which perferes ulaw.
When my * box has to transcode alaw to ulaw the sound get's one way choppy.
(alaw = ulaw is choppy, ulaw = alaw is fine).
I managed to fix the issue by forcing my SIP phone to use alaw only,
Hi there
I europe alaw is usual. I have a SIP Phone which perferes ulaw.
When my * box has to transcode alaw to ulaw the sound get's one way choppy.
(alaw = ulaw is choppy, ulaw = alaw is fine).
I managed to fix the issue by forcing my SIP phone to use alaw only, but is
this a know issue with
after recompilling asterisk (trunk-r75109) after system (mandriva
cooker) update (new glibc 2.6, gcc 4.2.1),
sound starts very choppy, when codec translation is performed,
if translation isn't needed, it sounds OK
any idea? until update, everything worked fine.
I'm using ztdummy as clock source.
Paco Brufal wrote:
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not with Playback() or Background()...
Do you
Hello,
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not with Playback() or Background()...
On Fri, 8 Jun 2007, Paco Brufal wrote:
Hello,
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not
Hi Everyone,
Google IS my friend. I found the solution via Google on the second
glance ;-)
It seems that the USB latency was too high and you had to increase a
CAPI-Buffersize in chan_capi.h:
#define CAPI_MAX_B3_BLOCK_SIZE 500
(German instructions:
Hi everybody,
I have a problem which I cannot eliminate on my own. Has anybody any idea
for the following:
I am using the asterisk-version from Debian-Testing (1.2.13) with the
latest chan_capi (also tried an older version).
When using the Capi-Channel, everything works fine except from the
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or
without ztdummy device, renice -20 on asterisk process and even real-time
priority on the host Windows XP box for the vmware process. I am
Hi,
I have lately noticed that we sometimes get choppy sound when recieving
calls from the PSTN (on a TE410P-card) that get sent to an external SIP
extension (over the internet) who has a somewhat bad connection.
The strange thing is that it still sounds good when calling internally
to the
PROTECTED]
Skickat: den 21 augusti 2006 15:15
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?
Hi,
I have lately noticed that we sometimes get choppy sound when recieving calls
from the PSTN (on a TE410P-card) that get sent to an external SIP
While the gurus are sleeping, I'll ask a few questions to get started...
I'm assuming this is using some kind of Voip, correct? If not, can you let us know what cards are involved? What kind of phone? Is it IP or analog?
But, to be honest, none of that should hinder performance. What
While the gurus are sleeping, I'll ask a few questions to get started...
I'm assuming this is using some kind of Voip, correct? If not, can you
let
us know what cards are involved? What kind of phone? Is it IP or analog?
I use sip protocol with broadvoice.
But, to be honest, none of
Hello,
I created this setup,
DSL--LINUX ROUTER---ASTERISK
Linux acts as router and forwards packets only
512M and AMD 1599.987 MHz
Asterisk
512M
AMD 2000 MHz
When I ssh to linux router during the call and
execute any command that requires cpu , then sound gets choppy.
Simple test
Title: Choppy Sound on PSTN End
Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux.
Our call routing is like this:
SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice
] On Behalf Of Tim Chandler
Sent: 02 May 2005 17:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Choppy Sound on PSTN End
Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz
processor. I am running the latest build of White Box Enterprise Linux.
Our call
can
expect less than ideal audio. Also, why disable GSM ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
ChandlerSent: Monday, May 02, 2005 11:23 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Choppy Sound
on PSTN End
Hi all,
I recently set up
Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler
Sent: 02 May 2005 17:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Choppy Sound on PSTN End
Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz
processor. I am
: RE: [Asterisk-Users] Choppy Sound on PSTN End
I have the same problem on a Dell 1850 with a TE410P and have been
attempting to narrow it down. Interrupts don't seem to be a problem and I
have two PRIs from two different suppliers and both have the same
static/chop on the line so it's not the PRI
Of Tim Connolly
Sent: Monday, May 02, 2005 11:22
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Choppy Sound on PSTN End
I have the exact setup
you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some
settings. You might turn on qualify
: RE: [Asterisk-Users] Choppy Sound on PSTN End
I doubt it is the RAID controller since my Dell server isn't using one and I
have this problem...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aza
Sent: Monday, May 02, 2005 11:09 AM
To: 'Asterisk Users
I'm using X-Pro
connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15
minutes in a call I get a lot of noise in my end. I don't think the other part
of the call hears it. After some 10 seconds or so everything is fine
again.
In my CLI I get
NOTICE[32322]: RTP
On Tue, Feb 01, 2005 at 10:37:36PM +0100, Anders F Eriksson wrote:
In my CLI I get NOTICE[32322]: RTP Transmission error to
85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN
through my X100P (clone) as well as call connected through my IP telephony
provider. I have
Hi All,
Whenever a call comes in via the ISDN and somebody leaves a voicemail,
the sound file recorded is very choppy. If I actually take the call, the
sound is not choppy so it's obviously something to do with the Asterisk
box itself having to do the recording. Perhaps the sound card drivers?
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