----- Original Message ----- From: "Julien Claassen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@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 _______________________________________________ -- 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
