Hello!
I'm currently using Asterisk 11 (due to the fact that Debian Wheezy has
Ast 11 in backports - so that I can have security updates from my
distribution).
I recently played a little with Asterisk to be able to hear
internet-radios over the local phones in my home. So I set up a couple
of mus
Hi,
What is the expected behavior when setting the MOH Class in the Application
MusiConHold? For simplicity purposes my dialplan looks like this:
Exten => moh,1,MSet(SIP_CODEC=ulaw,CHANNEL(musicclass)=ABC)
Exten => moh,n,Answer
Exten => moh,n,MusicOnHold(DEF)
Exten => moh,n,Hangup
If you dial in
Man, I was hoping it was something like that. I did read the release notes; I
must have missed that part.
This should solve the problem, so thanks again.
Mike
On July 20, 2017 1:09:08 PM EDT, Richard Mudgett wrote:
>On Thu, Jul 20, 2017 at 11:50 AM, mdiehl
>wrote:
>
>> I recently upgraded As
On Thu, Jul 20, 2017 at 11:50 AM, mdiehl wrote:
> I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that
> music on hold isn't working like it used to.
>
> It seems that even though the correct MoH class is being set, the system
> still plays the "default" music.
>
> All of my ca
I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music
on hold isn't working like it used to.
It seems that even though the correct MoH class is being set, the system still
plays the "default" music.
All of my call handling is done with an AGI script. When a call is made
Hi Jonas
Wouldn't this do the job?
touch /etc/asterisk/musiconhold.conf ; asterisk -rx 'module reload
res_musiconhold.so'
Pete
> On 31/03/2017, at 8:55 am, Jonas Kellens wrote:
>
>
> I would not know how to automate this through script...
>
smime.p7s
Description: S/MIME
Hello
I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open
with vi and close again) and then issuing a 'module reload
res_musiconhold.so' on the Asterisk CLI makes the new files load into
Asterisk.
Very strange !!
I would not know how to automate this through script...
> Hello
> as you can read in my original post "moh reload" and "module reload
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??
Just spotted that you are using Asterisk 1.8.32.3.
The bug I'm think
> Hello
> as you can read in my original post "moh reload" and "module reload
> res_musiconhold.so" does nothing.
> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??
I think there is currently a bug with MOH. For now, if you add a fil
Hello
as you can read in my original post "moh reload" and "module reload
res_musiconhold.so" does nothing.
Only at restart the new files are available.
Is this a bug ?? How can I get more debugging for this problem ??
Kind regards.
On 23-03-17 22:54, Administrator TOOTAI wrote:
Le 23/0
Le 23/03/2017 à 20:17, Jonas Kellens a écrit :
Hello
is there any more information on how to reload/read musiconhold files ?
CLI> module reload res_musiconhold
--
Daniel
On 07-03-17 10:46, Jonas Kellens wrote:
Hello
I did not mention it but of course the MOH directory is listed in
/etc/a
Hello
is there any more information on how to reload/read musiconhold files ?
Kind regards.
On 07-03-17 10:46, Jonas Kellens wrote:
Hello
I did not mention it but of course the MOH directory is listed in
/etc/asterisk/musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/
Hello
I did not mention it but of course the MOH directory is listed in
/etc/asterisk/musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
[myfolder_1]
mode=files
directory=/var/lib/asterisk/moh/myfolder/1
sort=alpha
[myfolder_2]
mode=files
directory=/var/lib/asterisk/moh/
Your new file is in the 'myfolder/1'' subdirectory of the MOH directory.
Either move the file into the MOH directory or define a new class in
musiconhold.conf that is for your directory.
On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens
wrote:
> Hello
>
> using Asterisk 1.8.32.3
>
> Current music
Hello
using Asterisk 1.8.32.3
Current music on hold :
myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_ca
Hi,
When using realtime MOH and an external script such as madplay if in
MusicOnhold we have set cachertclasses=yes, after an Asterisk restart the
first time a call comes in madplay is started. If ten people are using the
same moh class there is only one instance of madplay. The issue is that if
n
Is the subdir "Horaires" readable/executable for User Asterisk/Asterisk?
Did you try to convert it to wav?
Am 17.06.2013 09:47, schrieb Thorsten Göllner:
Take a look here:
http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/
Am 16.06.2013 09
If I read your log entries correctly, you are not playing any MOH at all. BackGround() normally
plays sound files from the language dependent sound directory.
jg
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i don't think's that it's the same problems, because me format_mp3.so is
loaded:
[root@ipbx Conf-Extensions]# asterisk -rx 'core show file formats'
Format Name Extensions
-- --
slin mp3mp3
gsmgsmgsm
slin192sln192 sln192
sl
Take a look here:
http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/
Am 16.06.2013 09:43, schrieb Olivier CALVANO:
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow 11.4.0
i use the package centos, i can't use menuselect no ?
but i think's that is loaded:
ipbx*CLI> module load format_mp3.so
Unable to load module format_mp3.so
Command 'module load format_mp3.so' failed.
[Jun 17 04:56:42] WARNING[8910]: loader.c:892 load_resource: Module
'format_mp3.so' already exist
On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO wrote:
>
>
> Hi
>
> we have a small problems.
>
> We have a Asterisk 1.6.1 old server with music on old.
>
> we have updated to AsteriskNow 11.4.0
>
> and now, when we want play sound, we have a errors:
>
> -- Executing [334xx@Accueil_HNO:2]
> B
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow 11.4.0
and now, when we want play sound, we have a errors:
-- Executing [334xx@Accueil_HNO:2] BackGround("SIP/SIP05-000c",
"Fermeture") in new stack
[Jun 16 07:35:06]
Look at asterisk 11
A option was added to play announcements between music Files and so forth
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.@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of
Adam Moffett
Sent: Monday, January 21, 2013 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MoH with message on intervals
I'm talking to somebody w
gt;
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam
>> Moffett
>> Sent: Monday, January 21, 2013 4:07 PM
>> To: Asterisk Users Mailing List
um.com] On Behalf Of Adam Moffett
> Sent: Monday, January 21, 2013 4:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] MoH with message on intervals
>
> I'm talking to somebody who wants to have a recorded message play
> periodicall
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Monday, January 21, 2013 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MoH with message on intervals
I'm talking to somebody who
I'm talking to somebody who wants to have a recorded message play
periodically for people on hold.
An example would be interrupting the hold music every two minutes to
play a message with business hours and current specials.
Seems like you could fake it by breaking the music files into two mi
non-zero on
'SIP/gw1-0066' in macro 'stdexten'
== Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-0066'
-- SIP/gw1-0069 answered SIP/6908-0068
-- Packet2Packet bridging SIP/6908-0068 and SIP/gw1-0069
Thanks,
Kevin Ora
: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some
reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently runni
vail.slin' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/gw1-05d6' in macro 'stdexten'
== Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-05d6'
Thanks!
Kevin Oravits
From: Danny Nic
-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450
phones.
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones.
With one of my sites, we're having an issue where when a call is transferred,
the MOH is not playing and all the caller is hearing is silence. The caller of
course thinks they have been hung up on, but the
-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
> Sent: Thursday, July 28, 2011 3:44 PM
> To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users
From: "Mike"
Sent: Thursday, July 28, 2011 1:29 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] MoH - conversion command
>
> > I should have said I am trying this bo
From: "Mike"
Sent: Thursday, July 28, 2011 1:29 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] MoH - conversion command
>
> > I should have said I am trying this bo
you have to select musical compositions that are 'less incompatible'
> with G.729's compression methods. When we chose the current MOH selections
> included with Asterisk, our initial list was much larger, but we had to
> remove some because they sound terrible when compressed with G.729.
>
Thank
On 07/28/2011 01:29 PM, Mike wrote:
I should have said I am trying this both from a landline using ulaw,
and from a Polycom phone using g729 codec. G729 is noticeablty worst,
as you`d expect, maybe this is what is reported by my customers.
Is there any way to have a "decent" g729 file, or sho
>
> > I should have said I am trying this both from a landline using ulaw,
> > and from a Polycom phone using g729 codec. G729 is noticeablty worst,
> > as you`d expect, maybe this is what is reported by my customers.
> >
> > Is there any way to have a "decent" g729 file, or should I just give
>
On 07/28/2011 11:57 AM, Mike wrote:
I should have said I am trying this both from a landline using ulaw, and
from a Polycom phone using g729 codec. G729 is noticeablty worst, as you`d
expect, maybe this is what is reported by my customers.
Is there any way to have a "decent" g729 file, or shou
> On 07/28/2011 10:53 AM, Mike wrote:
> > Hi,
> >
> > I've been trying to get MoH files to sound decent. I've got a hold of
> > Royalty-free Classical music (a safe choice for most of my customers)
> > and I`ve been trying to convert them to the normal telephony/Asterisk
> > format using sox. Unfor
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MoH - conversion command
On Thu, 28 Jul 2011, Mike wrote:
> I?ve got a hold of Royalty-free Classical music (a safe choice for
> most of my customers) and I`ve been trying to convert them to the
> normal tele
On Thu, 28 Jul 2011, Mike wrote:
I’ve got a hold of Royalty-free Classical music (a safe choice for most
of my customers) and I`ve been trying to convert them to the normal
telephony/Asterisk format using sox. Unfortunately, it sounds really
bad.
I convert files using:
sox "${INPUT
On 07/28/2011 10:53 AM, Mike wrote:
Hi,
I’ve been trying to get MoH files to sound decent. I’ve got a hold of
Royalty-free Classical music (a safe choice for most of my customers)
and I`ve been trying to convert them to the normal telephony/Asterisk
format using sox. Unfortunately, it sounds rea
Hi,
I've been trying to get MoH files to sound decent. I've got a hold of
Royalty-free Classical music (a safe choice for most of my customers) and
I`ve been trying to convert them to the normal telephony/Asterisk format
using sox. Unfortunately, it sounds really bad. I don't expect concert h
any reply on below issue..
Nikhil
On 06/06/2011 07:23 PM, Nikhil wrote:
Hi
when I Upload MOH file from Asterisk GUI ,it is getting success and
even not getting any error,But if check the destination path the file
is not showing , even the source file and destination path and formate
are co
Hi
when I Upload MOH file from Asterisk GUI ,it is getting success and
even not getting any error,But if check the destination path the file is
not showing , even the source file and destination path and formate are
correct .I am not getting any error log from asterisk console too. I
read in
On Fri, 3 Jun 2011, Nikhil wrote:
I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is not
working .help me on this
Not unless you provide some details.
What's not working? Is the file not being uploaded? Is the file not being
uploaded to the correct directory? Is the file in
Hi am
I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is
not working .help me on this
Thanks
Nikhil
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Good morning.
I have a problem with the moh
when I receive an external call, answer and transfer telephone call.
when I transfer the user does not feel the MOH.
if you transfer calls between the internal functioning properly moh
In the users.conf i have
mohinterpret=musiconhold
Kan you help
Hi Selby,
First of all thanks for reply, I added that line at the end of dialplan but
still the same.
Actually *asterisk default MOH is also not playing*. after 12 sec I get this
message...
* -- Stopped music on hold on SIP/1001-000f*
Strange when I did the same thing with asterisk 1.4.39 th
Add exten => 6000,n,StartMusicOnHold() to the end of your current dialplan and
try again.
Thanks,
--Warren Selby, dCAP
On Apr 8, 2011, at 1:51 AM, virendra bhati wrote:
> I am using Elastix. Asterisk is used for PBX application in Elastix. I want
> to make customize MOH. But not able to use
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working
*Asterisk Version 1.6
g List - Non-Commercial Discussion
> *Subject:* [asterisk-users] MOH on DAHDI PRI Channels
>
>
>
> Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
> connected with it. When the called party press hold on his phone then
> asterisk start MOH??
>
>
&g
_
[Shariq Khan]
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??
[Danny Nicholas]
Question #1
Dial(DAHDI/1/5551212,20,m) will play moh until the other end
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shariq Khan
Sent: Thursday, April 07, 2011 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MOH on DAHDI PRI Channels
Is it
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??
--
Regards,
Shariq Khan
0333-3501125
--
_
-- Bandwidth and Coloc
Hi,
I have some customers complaining about moh quality, but never about voice
quality. The moh files themselves sound great when listen to from a media
player, and given that I am streaming the files through DAHDI (while voice
goes from DAHDI to SIP, so more chances of problems) I would expec
@lists.digium.com
Subject: Re: [asterisk-users] MOH and parking
On 11-01-21 08:52 AM, Andrew Thomas wrote:
> I know that the 'fix' has just been applied
> (https://issues.asterisk.org/view.php?id=18262) - but why does it stop
> the moh only to start it again? This, also, s
On 11-01-21 08:52 AM, Andrew Thomas wrote:
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
After speaking with Shaun and Russell, this is lik
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
-- Executing [7000@chambers:1] Park("SIP/2000-0008", "") in new
stack
== Parked SIP/20
On Jan 4, 2011, at 8:49 PM, Earl Terwilliger wrote:
> On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote:
>> On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
>>> Hi list,
>>>
>>> I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and
>>> am getting this error :
>>>
>>> WARN
On Tuesday, January 04, 2011 04:29:49 pm bakko wrote:
> Hi
>
> CLI> module unload res_musiconhold.so
>
> CLI> module load res_musiconhold.so
>
> or
>
> service asterisk restart
>
> Regards
>
> --
> _
> -- Bandwidth and Coloca
On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote:
> On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
> > Hi list,
> >
> > I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and
> > am getting this error :
> >
> > WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd(
On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory
[snip]
Have you installed mpg123 or some ot
Hi
CLI> module unload res_musiconhold.so
CLI> module load res_musiconhold.so
or
service asterisk restart
Regards
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Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory
with the default musiconhold.conf file. When I change musiconhold.conf to this:
[default]
mo
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 http://s
El 04/11/10 17:14, Tzafrir Cohen escribió:
> In the 'files' mode Asterisk plays the music separately for each
> channel. If you use mpg123 or any other streamer, there is a single
> stream per class.
>
A single stream per class sounds like good efficiency. Could you please
tell me what streamers c
Please ignore this message (wrong subject by mistake). Please see message with
subject "2nd network interface for RTP/media"
Thanks
Harel
--
Message: 2
Date: Mon, 1 Nov 2010 12:52:16 +0100
From: Harel Cohen
Subject: [asterisk-users] MoH and stuch c
Hi All,
I would like to separate the media traffic from the signalling.
Can Asterisk send and receive media (rtp) traffic from a secondary network
interface?
Thanks,
Harel
This electronic message and any files transmitted with it are confidential and
intended sol
/2010 16:57, Stefan Schmidt escreveu:
> Danny Nicholas schrieb:
>> *From:* asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario
>> Quiroz
>> *Subject:* [asterisk-users] MOH in the middle of the call
>>
>
Danny Nicholas schrieb:
>
> *From:* asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario
> Quiroz
> *Subject:* [asterisk-users] MOH in the middle of the call
>
> >Hi, I have a very strange problem. In the mi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dario Quiroz
Subject: [asterisk-users] MOH in the middle of the call
>Hi, I have a very strange problem. In the middle of the call the MOH starts
for 30 seconds approximately.
>
Hi, I have a very strange problem. In the middle of the call the MOH starts
for 30 seconds approximately.
After this the call run normally.
Anybody have an ideia or has some similar problem?
Thanks in advance!!
--
Atenciosamente,
---
>> Something you may want to try (its fixed it for us) is putting an I
>> (uppercase I) on the asterisk invocation line.
>>
>> We run servers in the cloud and can't get reliable timing from ISDN
>> cards etc so this instructs asterisk to generate its own internal
>> timing. If you have ISDN you pro
> Something you may want to try (its fixed it for us) is putting an I
> (uppercase I) on the asterisk invocation line.
>
> We run servers in the cloud and can't get reliable timing from ISDN
> cards etc so this instructs asterisk to generate its own internal
> timing. If you have ISDN you probably
Something you may want to try (its fixed it for us) is putting an I
(uppercase I) on the asterisk invocation line.
We run servers in the cloud and can't get reliable timing from ISDN
cards etc so this instructs asterisk to generate its own internal
timing. If you have ISDN you probably don't want
>> I have been facing an issue that voice is getting distorted sometimes in
>> MOH
>> (MusicOnHold) application.
>> I have checked and confirmed that lame is properly installed, even tried
>> native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
>> can't be eliminated.
>> I came
Op 21-07-10 18:05, Danny Nicholas schreef:
>
>> QOS on ISDN? Don't know how to do that.
>
>> Ron
>
>> NeoNova BV
>> innovatieve internetoplossingen
>
> It's not the ISDN part, it's the internal SIP part controlled by the
> network. Most calls on Asterisk consist of two or more legs; one or more
Op 21-07-10 17:18, Danny Nicholas schreef:
>
> This sounds like a QOS issue (quality drops during heavy usage). Since it
> was more prominent when you were on ISDN, that pretty much verifies it for
> me. Can you prioritize voice traffic?
>
>
QOS on ISDN? Don't know how to do that.
Ron
--
NeoN
This sounds like a QOS issue (quality drops during heavy usage). Since it
was more prominent when you were on ISDN, that pretty much verifies it for
me. Can you prioritize voice traffic?
--
_
-- Bandwidth and Colocation Provi
Op 21-07-10 08:32, MohammedShehzad schreef:
> Hello,
>
> I have been facing an issue that voice is getting distorted sometimes in MOH
> (MusicOnHold) application.
> I have checked and confirmed that lame is properly installed, even tried
> native formats (ulaw, alaw, gsm), but the randomly seen dis
On 21 July 2010 10:59, MohammedShehzad wrote:
>> > I have been facing an issue that voice is getting distorted sometimes in
>> > MOH
>> > (MusicOnHold) application.
>> > I have checked and confirmed that lame is properly installed, even tried
>> > native formats (ulaw, alaw, gsm), but the randoml
> > I have been facing an issue that voice is getting distorted sometimes in MOH
> > (MusicOnHold) application.
> > I have checked and confirmed that lame is properly installed, even tried
> > native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
> > can't be eliminated.
> > I c
On Wed, Jul 21, 2010 at 12:02:03PM +0530, MohammedShehzad wrote:
> Hello,
>
> I have been facing an issue that voice is getting distorted sometimes in MOH
> (MusicOnHold) application.
> I have checked and confirmed that lame is properly installed, even tried
> native formats (ulaw, alaw, gsm), but
Hello,
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
can't be eliminated.
I came to know abou
I have defined a moh class in musiconhold.conf and set the class using
SetMusicOnHold in my dialplan:
; musiconhold.conf
[music1]
mode=files
directory=moh/music1
sort=alpha
; extensions.conf
exten => s,n,SetMusicOnHold(music1)
exten => s,n,Queue(${ARG1})
Inside /var/lib/asterisk/moh/music1 there
On 03/06/2010 12:46 PM, Joseph wrote:
> I have bragged/connected two asterisk over IAX2
The IAX2 stack has a built-in preference for meekness and modesty.
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web: http://www.evaristesys.com/
-
I have bragged/connected two asterisk over IAX2 but MOH is not working when I
try to dial directly between them?
MOH is working over SIP but not IAX2 ??? (the two asterisks are connected over
VPN).
Here is the log:
-- Executing [...@extensions:1] Dial("IAX2/home_server-1756",
"SIP/877|25|m(defa
Tilghman,
That's what I finally figured out... my understanding, though, is that it is
preferred to use 'file' over mpg123?
On Sat, Mar 6, 2010 at 1:32 AM, Tilghman Lesher wrote:
> On Friday 05 March 2010 17:19:06 Matt wrote:
> > For some reason I have to set the type to 'files' if I set it to
>
On Friday 05 March 2010 17:19:06 Matt wrote:
> For some reason I have to set the type to 'files' if I set it to 'quietmp3'
> I get nothing, even though the files are valid MP3 files that play on
> another asterisk system... does that mean I've got something installed
> wrong?
Mostly likely you did
Un-top-posting...
>> On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote:
>>>
>>> I'm trying to setup my asterisk system for the least overhead as
>> possible.
On Fri, 5 Mar 2010, Matt wrote:
> For some reason I have to set the type to 'files' if I set it to 'quietmp3'
> I get nothing, even though the
For some reason I have to set the type to 'files' if I set it to 'quietmp3'
I get nothing, even though the files are valid MP3 files that play on
another asterisk system... does that mean I've got something installed
wrong?
2010/3/5 Håkon Nessjøen
> On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote:
On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote:
>
> I'm trying to setup my asterisk system for the least overhead as possible.
>
> My understanding (and experience with other systems) leads me to believe I
> can run any MOH using a certain class through a single 'player' as opposed to
> starting an
Forgot to include: I'm running 1.6.2.5
On Fri, Mar 5, 2010 at 5:29 PM, Matt wrote:
> I'm trying to setup my asterisk system for the least overhead as possible.
>
> My understanding (and experience with other systems) leads me to believe I
> can run any MOH using a certain class through a single
I'm trying to setup my asterisk system for the least overhead as possible.
My understanding (and experience with other systems) leads me to believe I
can run any MOH using a certain class through a single 'player' as opposed
to starting an independent stream for each MOH instance. However, try as
Hi All,
I have issue with moh in asterisk 1.4 branch, (1.4.26)
I have created two different moh classes and it is fine,
I have assigned 1st one to sip user 1001, in sip entries like
musiconhold=classname.
done same for 2nd one to sip user 2001,
And now i am making internal calls between then as nor
Peder wrote:
> I am having a strange problem with MOH. Say I have two users, A and B. I
> can set MOH in the extension for B and if A calls B and B hits hold, A will
> hear B's hold music. If however A hits hold, it goes to the default music.
> If I pull the setmusiconhold from extensions.conf a
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