[asterisk-users] jittery audio in voiceprompts
Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under vmware and both exhibited this issue. Linux box is perhaps a little under powered, it is an Intel Celeron 467Mhz. I tested with asterisk niced to -18, which did not change the problem. I am using a Linksys SPA942 sip phone, using ALAW. I have tried gsm and alaw prompts but that didn't solve it. I am running Linux blue 2.6.16.20 on a debian stable machine. I have been compiling and installing asterisk from source. I have tried looking at the debug messages in asterisk but nothing seems to indicate an issue. I read somewhere that disabling X can help, but it did not in my case. I am at a loss as to how I might track down the problem and fix it. Any pointers would be greatly appreciated. Thanks, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jittery audio in voiceprompts
Steve Murphy wrote: On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote: I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. What do you have installed, that will provide the 1Khz timing interrupts you will need to function properly? Err.. I was not aware I would have to install anything to do that. I guess that could mean I have nothing installed. What should I have installed? Thanks, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jittery audio in voiceprompts
Michelle Dupuis wrote: Isn't there a zap dummy (or something that uses the RTC) included in Asterisk 1.40 that creates the timing source? We don't install any external timing sources and we don't have choppyness problems on pure sip connections... Yes, I have been looking into that after reading Steve's response. Unfortunately I get a compile error with it. I'll try a newer kernel. I have a pure SIP installation also Jason - is this on a standard PC motherboard (or a mini device like Linksys WRT)? Yes, standard PC (although older as mentioned in previous post) Thanks, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jittery audio in voiceprompts
Hi Florian Actually, I doubt the timing source will be required if you only use playback or background commands with the supplied gsm prompts. We run lots of machines without it. Timing sources are used for some cases of musiconhold, meetme and the likes, but not for regular stuff. what about for playing voiceprompts? Jason, if you do a 'vmstat 1' on the unix prompt when a call is run, does it ever hit an idle count of 0 somewhere ? If so, you have performance issues, if not, you'd probably look toward the network, or perhaps a silly Voice Activation setting in your phone. I tested vmstat, it only occasionaly reaches 0, usually it hovers around 99-100. I made sure the phone does not do silence detection. If possible, you could also try and look at a tcpdump capture of your traffic using wireshark to see if there is specific jitter or packetloss in the audiostream as it leaves the server. Thanks, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
