OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there.
TIA, Todd ----- Original Message ----- From: "Brian West" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Monday, September 03, 2007 6:10 PM Subject: Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly > Try setting the RTP packets to 0.020 instead of 0.030 which is the > default on the SPA's > > /b > > On Sep 3, 2007, at 5:00 PM, Todd Reese wrote: > > > Hi all, > > > > I have just install and licensed Cepstral's Allison08kHz on my > > Asterisk > > 1.4.11 system. > > > > I can call the Allison's extension from my Grandstream IP Phone and > > she's > > clear as a bell, but when a call to her extension traverses through > > one of > > the Linksys/Sipura 3102 or 2002, she's got the jitters bad. > > > > The SPA-202 has only an extension phone on it and the SPA-3102 is > > my FXO > > from my Vonage Motorola box. > > > > > > Any clues where to start looking to clear this up? > > > > > > TIA, > > > > Todd Reese > > > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > 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-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
