Guys, The sounds i am playing back are .gsm files , but i will add the format=wav49|wav to voicemail.conf and see whats happens.
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. I stop it, when i checked the playback it was still low, and a piece of it was chopped off. Thanks again guys for your help. On Sun, Jul 13, 2008 at 8:25 PM, Steve Prior <[EMAIL PROTECTED]> wrote: > Try adding the following to your voicemail.conf context: > > format=wav49|wav > > Steve > > Leotis buchanan wrote: > > Hey Guys, > > > > I have configured my first asterisk box. it works ok so apart, but the > > playback sound quality is terrible, its low and the output sounds > > distorted and its seems to have been clipped. > > > > Can anyone help. > > > > > > > > > > > > On Sun, Jul 13, 2008 at 11:00 AM, Chris Rowson > > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > wrote: > > > > >> Hi, this is my first post to the list, but I have tried to search > > >> elsewhere for a solution > > >> <SNIP> > > >> I'm using sipgate.co.uk <http://sipgate.co.uk> for incoming > > calls, but when I make a test > > >> call from the PSTN, the call just dies without connecting to my > > >> Astlinux box. (I'm monitoring asterisk console via 'asterisk > > -rvvvvv' > > >> and see nothing). > > >> <SNIP> > > > > Thanks for the suggestions. I ran tcpdump and it indicated that > > traffic on that port was being forwarded to the asterisk server. It > > looks like I basically wrote a load of nonsense in the > extensions.conf > > file. I edited the file to input the extension the incoming call > > should be coming from and it now works. > > > > Working file --- > > > > [from-pots] > > exten => 277****,1,Answer() > > exten => 277****,n,Wait(3) > > exten => 277****,n,Playback(tt-weasels) > > exten => 277****,n,Hangup() > > > > So in summary it was basically me misconfiguring the box... > > > > Cheers > > > > Chris > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > -- > > Leotis Buchanan > > Manager/Electronic Design Systems Engineer > > Exterbox.com > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Leotis Buchanan Manager/Electronic Design Systems Engineer Exterbox.com
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
