Guys, <br><br>The sounds i am playing back are .gsm files , but i will add the format=wav49|wav to voicemail.conf and see whats happens. <br><br>I tried fxotune -i 4 , i was wondering since my fxo card is on channel 1, should it not be fxotune -i 1. Well , fxotune started ,but the tx and rx kept oscillating between 200 and about 217. It ran for ever. <br>
<br>I stop it, when i checked the playback it was still low, and a piece of it was chopped off. <br><br>Thanks again guys for your help.<br><br><div class="gmail_quote">On Sun, Jul 13, 2008 at 8:25 PM, Steve Prior <<a href="mailto:sprior@geekster.com">sprior@geekster.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Try adding the following to your voicemail.conf context:<br>
<br>
format=wav49|wav<br>
<br>
Steve<br>
<div class="Ih2E3d"><br>
Leotis buchanan wrote:<br>
> Hey Guys,<br>
><br>
> I have configured my first asterisk box. it works ok so apart, but the<br>
> playback sound quality is terrible, its low and the output sounds<br>
> distorted and its seems to have been clipped.<br>
><br>
> Can anyone help.<br>
><br>
><br>
><br>
><br>
><br>
> On Sun, Jul 13, 2008 at 11:00 AM, Chris Rowson<br>
</div><div class="Ih2E3d">> <<a href="mailto:christopherrowson@gmail.com">christopherrowson@gmail.com</a> <mailto:<a href="mailto:christopherrowson@gmail.com">christopherrowson@gmail.com</a>>> wrote:<br>
><br>
> >> Hi, this is my first post to the list, but I have tried to search<br>
> >> elsewhere for a solution<br>
> >> <SNIP><br>
</div>> >> I'm using <a href="http://sipgate.co.uk" target="_blank">sipgate.co.uk</a> <<a href="http://sipgate.co.uk" target="_blank">http://sipgate.co.uk</a>> for incoming<br>
<div><div></div><div class="Wj3C7c">> calls, but when I make a test<br>
> >> call from the PSTN, the call just dies without connecting to my<br>
> >> Astlinux box. (I'm monitoring asterisk console via 'asterisk<br>
> -rvvvvv'<br>
> >> and see nothing).<br>
> >> <SNIP><br>
><br>
> Thanks for the suggestions. I ran tcpdump and it indicated that<br>
> traffic on that port was being forwarded to the asterisk server. It<br>
> looks like I basically wrote a load of nonsense in the extensions.conf<br>
> file. I edited the file to input the extension the incoming call<br>
> should be coming from and it now works.<br>
><br>
> Working file ---<br>
><br>
> [from-pots]<br>
> exten => 277****,1,Answer()<br>
> exten => 277****,n,Wait(3)<br>
> exten => 277****,n,Playback(tt-weasels)<br>
> exten => 277****,n,Hangup()<br>
><br>
> So in summary it was basically me misconfiguring the box...<br>
><br>
> Cheers<br>
><br>
> Chris<br>
><br>
> _______________________________________________<br>
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><br>
><br>
><br>
><br>
> --<br>
> Leotis Buchanan<br>
> Manager/Electronic Design Systems Engineer<br>
> Exterbox.com<br>
><br>
><br>
</div></div>> ------------------------------------------------------------------------<br>
<div><div></div><div class="Wj3C7c">><br>
> _______________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Leotis Buchanan<br>Manager/Electronic Design Systems Engineer<br>Exterbox.com