Displaying 20 results from an estimated 30000 matches similar to: "Asterisk Sound Issue"
2007 Jun 08
3
choppy sound with playback, background, etc... but not with musiconhold
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()...
If I move app_playback.so from this system to another asterisk,
playback works fine...
Do you
2005 Mar 23
4
Playback of sound files but no sound
Hello,
I'm running asterisk-1.0.6 on a centos3.4 box.
I'm still in testing phase and so far everything is running smoothly.
I'm now trying to play a soundfile or an mp3file using 'MP3Player',
'Playback'
or the 'Background' commands, but don't get any sound.
The logfile says:
-- Executing BackGround("SIP/joa-9def", "tt-weasels") in
2006 Jan 19
1
Sound issue with Asterisk
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
======================
Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
anything in there that related to what you said. Or is it in a
different file or location?
I am
2008 Sep 11
1
Probably very simple... call a number and play a sound?
Hey hey...
I'd like to create a new feature code in asterisk so when a user dials...
say... *00, it would then call some other extensions and play a sound file
to them.
So far, this is what I have...
[testing-custom]
exten => *00,1,Wait(1)
exten => *00,2,Playback(beep)
exten => *00,3,Playback(beep)
exten => *00,4,AGI(festival-script.pl|I will now attempt the call)
exten =>
2004 Dec 09
6
Horrible MeetMe performance
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
conference, the audio is very delayed, choppy and segmented -- totally
unusable.
At the
2004 Jul 14
3
Voicemail/autoattendant not working
I'm pretty much a newbie to this but still think I've been around the
various help pages, voip-info.org etc to be fairly sure I'm not
missing something here so your help is appreciated!
I have a box running RedHat9 at home with the latest CVS of Asterisk
and all works fine.
At the office, we installed Gentoo linux on a machine, downloaded the
latest CVS of Asterisk, set it up. All
2004 Jan 08
1
Strange Call waiting problems - SNOM 200 & Grandstream Budgetone
Hi
I am setting up an Asterisk System in an office environment, Incoming and
Outgoing calls are working ok, but i am having a few strange problems
regarding call waiting.
With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call
comes in, the call waiting beep is played and the light flashes, but if you
hang up the 1st call, instead of the phone ringing, it connects the
2008 Mar 18
2
ztdummy problem causing playback () to fail
Hi, I am having problem with my Asterisk installation and find out it
has to do with ztdummy.
if the ztdummy module is loaded, the asterisk playback() command
will not play files. DTMF is still properly received. If the ztdummy
module is unloaded, sound playback works again.
Here is my version
zaptel-1.4.9.2
linux-source-2.6.18
asterisk-1.4.18
Can anyone tell me how to fix it? Or should I
2017 Jun 06
5
asterisk server - no sound
hello folks,
this might be a simple question...
I just installed asterisk in a debian server.
All seems to be running fine, but the audio sent by the server.
If I have one of my registered peers call and extension (102) that plays
back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
answers and prints no errors.
Its `sip show channels` prints:
Peer User/ANR Call ID
2007 Nov 02
3
ztdummy and BackGround
2008 Jan 30
4
Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is "cut".
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch =>
2008 Apr 03
6
ztdummy
What does it take to get ztdummy to work correctly?
I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel
1.4.9.2
Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
Problem is playback() does not work. So then I stop zaptel, asterisk
runs and playback() now
works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for
that.
I am
2010 Apr 22
4
More efficient dial plan for a list of selective inbound numbers
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.....)
[custom-inbound]
exten => _556,1,answer
exten => _556,n,playback(beep)
exten => _557,1,answer
exten => _557,n,playback(beep)
exten => _558,1,answer
exten => _558,n,playback(beep)
exten =>
2003 Nov 04
0
Need Help with SIP/H323.
Hi list,
why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)?
could anybody please give any idea to solve this issue?
Please, let me know.
Thanks in Advance.
N.B.
The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are:
***************************************
2007 Dec 03
4
Soundcard necessary on an asterisk server to get output of playback()??
Hi,
I' still fighting the problem, that I can talk from one SIP phone to
another, but I can't hear the output of the playback or similar
applications:
exten => 202,1,ANSWER()
exten => 202,2,PLAYBACK(tt-monkeys)
exten => 202,3,HANGUP()
When I dial 202, asterisk show the following on the cli:
-- Executing [202 at local:1]
2008 Feb 02
3
Zaptel timer on Intel Dual Core servers
Friends,
I'm having severe problems with zaptel timers on Intel Dual Core
systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM
or PRI cards - all ends up with large timer probems - zttest going
down to 50% accuracy on some systems, even to -1 on ztdummy systems
and voice quality is no more. A restart is the only way to get back
to a working system.
We're only
2007 Mar 12
2
Playback 5% Too Fast?
Hi All
I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application. There are clicks every few
seconds or more frequently that is audible on the remote end (PSTN), but
not on the Asterisk recording of the call. If I record the remote end
and compare it to the local recording, it appears to be about 5%-7% too
fast - i.e. if I synchronise the
2011 Jul 01
0
IVR sound after dial sip
Hi, I have a ivr, and I need to make a beep sound playback after
phone when to dial sip DIALSTATUS} = $ {ANSWER
example
1234,1,Answer()
1234,n,Dial(SIP/1234)
;When 1234 sip phone answer te
call, playback beep on this sip phone.
how could I do this?
thanks
for any help
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2005 Feb 04
2
No Playback() when Digicom TE110P enabled
I have a Digicom TE110p card installed in our exchange. I have compiled
and installed libpri, zaptel and recompiled and installed asterisk.
I have configured udev as I am running Fedora Core 3.
The problem that I have is that when zaptel is not running everything
works fine. However when I start zaptel (service start zaptel) then I
can make normal calls ok but the 'Playback()'