Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
On Sep 5, 2007 3:36 PM, Kai-Uwe Jensen [EMAIL PROTECTED] wrote: How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. Also fixed it here. I had some quite bad jitter on the first few seconds of speech with the default setting (app_swift-2.0rc1). Searched the Asterisk archives, found your message, made the change and voila! Thanks, Shaun ___ --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
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
Kai-Uwe Jensen wrote: How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. Can you specify exactly where you made this change? I'm looking at the source for app_swift-0.9 right now and don't see a framesize constant. I'm getting some breakup when using app_swift over an IAX connection and thought I'd try this. Thanks Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
Sure. Sorry to be unclear about it. I was using app_swift-2.0rc1, from http://www.mezzo.net/asterisk/app_swift.html. Part of that package is app_swift.c. At line 68, I changed the declaration const int framesize = 160*4; to const int framesize = 20; That fixed things here. As it seems, that fix may not work (or even be appropriate) for app_swift-0.9, which I would assume you got from loopfree.net. On 9/6/07, Steve Prior [EMAIL PROTECTED] wrote: Can you specify exactly where you made this change? I'm looking at the source for app_swift-0.9 right now and don't see a framesize constant. I'm getting some breakup when using app_swift over an IAX connection and thought I'd try this. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. On 9/3/07, Todd Reese [EMAIL PROTECTED] wrote: 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 -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --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
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
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 asterisk-users@lists.digium.com 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
