[Asterisk-Users] Prompts and sound quality of the X100P card(FXO card)

Lee Goodman lee.goodman at comcast.net
Wed Sep 10 09:36:54 MST 2003


Thanks for the info. Pretty much what I figured. I would be VERY interested
in hearing the prompts over a Digital T1 line instead of my analog port. If
you could make a phone number available, I would appreciate it very much

Thanks

Lee Goodman
lee.goodman at comcast.net


----- Original Message -----
From: "Steven Critchfield" <critch at basesys.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, September 10, 2003 11:31 AM
Subject: Re: [Asterisk-Users] Prompts and sound quality of the X100P
card(FXO card)


> On Wed, 2003-09-10 at 10:15, Lee Goodman wrote:
> > Hi
> >
> > We are trying to get better sound quality out of the prompts on our
Asterisk
> > system. We had some new ones made by thevoice.digium.com and they are in
WAV
> > format instead of the default GSM format on the Asterisk server. The
problem
> > is, when you dial in to the server using the FXO card (X100P) you really
> > can't tell the difference between the WAV prompt and the GSM prompt,
> > however, if you listen to the prompts via a SIP phone you can hear a
clear
> > difference. Is the FXO card compressing the WAV file? Is there anyway to
> > tweak the FXO card to make it sound better?
>
> Via SIP you are only doing 1 D/A conversion, and it is closest to the
> speaker. At worst on the SIP connection, the only change to the prompt
> is a codec change and that is unlikely.
>
> Via X100P, You have a D/A conversion to get it on the line, then at the
> phone switch it is digitized and dropped off at the appropriate port to
> go out to whatever phone you are using where there is another D/A
> conversion then it goes to your phone. If it is a cell phone, then there
> is compression on the cell link before it gets to you and the D/A
> happens in the phone itself. Not to mention the probability of a mile or
> more of analog copper in the link.
>
> So this isn't necessarily a problem with the X100P but in the underlying
> technology. If you like, I could make a phone number available for you
> to hear your prompts played over a PRI connection so you can see that it
> is just an analog problem.
> --
> Steven Critchfield  <critch at basesys.com>
>
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> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users




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