[Asterisk-Users] Music on Hold Quality

[Colin Anderson]

My IVR prompts I create by mixing together the voiceover with an imported
music MP3 in Audacity. I use the tools in Audacity to remove the hiss,
normalize the audio, and bump up the low end (that really helps) I save it
as CD-quality stereo then downsample it to 16 bit, 8Khz mono with good old
sndrec32.exe in Windows. Copy it over to Asterisk and sox it as GSM files,
and the sound is perfect, no skips, hiss or anything like that. The system
is under light load (maybe 5-15 calls concurrent at any one point) ,
however, so I can't comment on if it's an Asterisk thing for it to skip
under heavy load or what. 

Works for me, though. 

-----Original Message-----
From: Jorge Alayon [mailto:j.alayon at ses.com.ar]
Sent: Wednesday, September 28, 2005 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Music on Hold Quality


Music on hold audio should be resampled to 8 bits  16 Khz mono and
preencoded, so audio distortion is minimized in Asgerisk encoding. This
theory has worked form me on other commercial platforms, but not yet on
Asterisk, because MP3s cannot be resampled that way. If anyone figures it
out, please advice.

Regards,

Jorge Alayón

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]En nombre de Matt
Enviado el: Miércoles, 28 de Septiembre de 2005 06:42 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Music on Hold Quality


I have heard this issue when on hold with Cisco and Vonage... I don't
think it's an asterisk problem I htink it's a G711 "problem"... or gsm
"problem".    Basically they are made for voice, and I think the music
goes outside their encoding ranges... sound logical?

On 9/28/05, canuck15 <canuck15 at hotmail.com> wrote:
>
>
> I was on hold at Digium and noticed they had that EXACT problem.  It was
> REAL bad on the one occasion I was on hold with them a couple weeks ago.
If
> Digium has it then there must be some inherent issues with Asterisk that
> need to be worked out.  I personally think this should be given a HIGH
> priority.
>
> Just my opinion as I am not a coder and cannot contribute in that way.
>
>
>  ________________________________
>  From: Justin Selleck [mailto:jselleck at smoothfusion.com]
> Sent: Wednesday, September 28, 2005 11:19 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Music on Hold Quality
>
>
>
>
>
> Does anyone know how to maximize music on hold quality on calls inbound
from
> PSTN?   I know that it is common to have choppy and static sounding music
on
> hold when connecting via PSTN but how can that be minimized?  I assume
that
> the bitrates, type of music, etc can minimize the effects.  Does anyone
have
> any experience in this area?  Do you know where I should look for more
> information?
>
>
>
> Thanks!
>
>
>
> -Justin
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