[Asterisk-Users] Re: wav instead of gsm for vm-sounds?

[Damon Estep]

> In article <432B744E.8040304 at digium.com>,
> Kevin P. Fleming <kpfleming at digium.com> wrote:
> > Damon Estep wrote:
> > > Do you simply replace the .gsm files with .wav files and it plays
them
> > > in these apps, or is there more to it?
> > >
> > > I am talking about the built in functionality of vm, queues,
agents --
> > > not the playback app.
> >
> > Every attempt to play a file in Asterisk (that doesn't specify the
> > extension) will use the 'best' format it can find for the call's
format.
> > The applications that ship with Asterisk do not specify any
particular
> > format, so your best bet is to provide the files in all the formats
you
> > expect your callers to be using, so Asterisk never has to transcode
a
> > file playback. The same thing is true for voicemail recording; you
> > should record in every format your users might call in with, unless
the
> > extra disk space consumption would be an issue.
> 
> Do wav or sln versions exist of the standard Asterisk sounds by
Allison?
> I mean the versions before GSM compression was applied, not just ones
> obtained by uncompressing the GSM again.
> 
> Cheers
> Tony
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
> _______________________________________________

Not sure about that Tony, we recorded a full set of sounds on our own so
we would have a consistent voice in our customizations. We also change
all of the references from 'extension' to 'phone number'

It took a little time, but well worth it.

We used a creative labs audigy sound card(good SNR) on a windows machine
with a good mic and then processed the sounds in adobe audition,
applying a high pass filter, low pass filter, silence removal, sound
level normalization, and a 10% time reduction. The 10% time reduction
and silence removal really helps things sound professional.

We started with 16bit, 32khz, mono pcm wave files, we then did our
processing, saving the file as 16bit, 8khz, mono after the processing
and then finally converting to ulaw files using sox and a script found
on the wiki with a few changes.

32khz gives good resolution for processing, and scales to 8khz well as
it is a multiple of 8khz. 44khz is trickier to scale to 8khz and may
result in some artifacting. 8khz is not a high enough sample rate to get
good filter processing with some of the adobe filters.

Here is the script I used in sox to do the final conversion;

# for a in *.wav; do sox $a `echo $a|sed -e s/wav//`ul ; done

Thanks to Kevin flemming for pointing me in the right direction on this.