[asterisk-users] Audio Files

[Abel Monzon]

----- Original Message ----- 
From: "Julien Claassen" <julien at c-lab.de>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Friday, September 26, 2008 8:03 PM
Subject: Re: [asterisk-users] Audio Files


> Hi!
>   I think all - at least all PSTN - calls have the same quality in means 
> of
> bitrate, number of channels and samplerate.
>   It's 8kHz, 16bit and mono.
>   About noise, I didn't have problems with that. Seems it's not really 
> about
> "quality". Probably it would be helpful, if you tell us, which
> extensions/protocol you used.
>   Kindest regards
>           Julien
>
>

Well, I had installed the sample with gmake, and I add my own extension,

exten => 269544,1,dial(Sip/user1,20)
exten => 269544,2,hangup()
and
exten => 269544,1,dial(Sip/user2,20)
exten => 269544,2,hangup()


exten => 1,1,Playback(Wellcome)
exten => 1,2,hangup()

So, When I call from user1 to user2, have noise, If I call from user1/user2 
to extension 1 the Playback have noise to. but, If I call to inexitent 
extension like 0000 the asterisk reproduced a error sound and not have 
noise..

What's is wrong??

Abel