search for: stutteri

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls - very stuttery , have tried a few different codecs. I assume this is a granular clocking issue , and wondering if anyone has anything I could try to fix or mitigate the problem in ESXi environment . We have

I thought the same thing before I made my reply but zapata.conf seems to be the only config file that deals with echo at all. From what I understand of 'echotrain' is that at the beginning of the call it sends a short signal out that measure echo in an attempt to try and cancel it. I was wondering if you tried using it and if so was it of any help? Sincerely, Steve But is Zapata.conf

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

I am just messing around with wine and World of Warcraft, I am behind a firewall and presently unable to get out to WoW, but can get to the login screen fine. The problem is that I can run it with a clean .wtf (config file) and it starts up ok with defaults (DirectX mode) and seems smooth, I exit and then launch it again and it seems to "stutter" and I dont mean sound, but graphically.

Strange issue that I can't figure out and I am hoping someone may have some ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it runs like a top and I am not going to mess with it). *B rsyncs config from *A. *A dies. I bring up *B and it all works fine, except for one issue. Calls to voicemail are garbled and low. Phone to phone and phone to gateway work perfect. If I

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

Greetings - I''m new to QoS, so please be gentle (and yes, I''ve RTFM, though I don''t understand every bit of it) Here''s the thing; I''ve tried several scripts--simple and complex--for classifying my Vonage traffic into a high-priority queue, but no matter what I do it doesn''t seem to work. Right now I''m using the HTB script

Folks, I've been running an icecast server for about a year, and never really had the chance to test it out on the other side of our Internet link. We've had a lot of people trying us out, but very short connect times in the whole. I finally got a cable modem at home, tried having a listen, and found the stream to be very stuttery. So, first things first, I went and grabbed the

https://bugs.freedesktop.org/show_bug.cgi?id=90453 Bug ID: 90453 Summary: Desktop periodically freezes when using Nouveau & getting "nouveau E[PDISP]" errors in dmesg Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: