similar to: Switch to Multi-Proc -> Choppy sound?

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Any idea what causes the warning "Unable to open pseudo channel for timing... Sound may be choppy."? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated!

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

http://bugs.freedesktop.org/show_bug.cgi?id=19986 Summary: Nouveau (20090116) fails to load on a PowerPC Pegasos 2 (OF) using a Quaddro FX 1100 (AGP) Product: xorg Version: 7.4 Platform: PowerPC OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component:

Hello, I created this setup, DSL------LINUX ROUTER-------ASTERISK Linux acts as router and forwards packets only 512M and AMD 1599.987 MHz Asterisk 512M AMD 2000 MHz When I ssh to linux router during the call and execute any command that requires cpu , then sound gets choppy. Simple test would be establish a call and start "du /" on the router. The same applies to asterisk box.

after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ?ast_context_add_ignorepat2?:

Andrew, Please find links below to sound clips from different combination. You will see that the 2.6.9.34 kernel on trixbox 2beta still does not sound ok. What did you do on the trixbox 1.2.2 vmware load to make it work better? I say better because it is still not perfect as you will notice that it accelerates near the end. I really appreciate your help and the great work you are doing. Also,

Hi there, I''ve just installed login_engine and user_engine from the repositories, followed the setup procedures as documented in: vendor/plugins/user_engine/README But I''m getting the following failure: vince@vaio:~/Projects/Booking$ rake bootstrap (in /home/vince/Projects/Booking) rake aborted! undefined method `edit'' for class `UserController''

I have gotten the Cisco 7960 working with my Asterisk system under SIP. The version is 5.03 that I am using. Cisco Support said I should not upgrade to version 6 yet. My next question is the sound is patchy when people here me. But I can hear them just fine not patchy. I have the 188 page Admin manual and it seem not to say anything about improving the sound. All other phones like IPDialog work

I asked about 18 months ago if gallery2 was available in any of the CentOS repositories, and it seemed that at that time it was not. However, there was some talk of putting it in epel or epel-testing. But I looked just now, and did not find it there. I actually installed gallery2 on my server by following the instructions at <http://codex.gallery2.org/Gallery2:Preinstaller>. But I've

I've setup a trial Asterisk install based on the RH8 install guide mentioned on this list. (Thanks, Andy!) I've configured two other working systems with SJPhone software for SIP. However, while I can call the phones, the communication is choppy. About every three or four seconds it cuts out briefly and then returns to normal. Perhaps someone can give me some pointers. My setups are

Hello, i have an ubuntu 8.04 LTS Virtual Server with Dovecot 1.0.10 and postfix. Now i tried to update Dovecot because i need Quota, which is not possible in 1.0 but in 1.2 So i added the packages to my /etc/apt/sources.list: deb http://codex.xiaoka.com/apt hardy main restricted deb http://archive.ubuntu.com/ubuntu hardy-backports main universe multiverse restricted and updated dovecot with

hi, I want to share my odysse from yesterday until now with someone who might help me. (my real problem starts after the mass effect 3 part) So yesterday I bought MassEffect 3 after reading the topic on it http://appdb.winehq.org/objectManager.php?sClass=version&iId=25500 and http://appdb.winehq.org/objectManager.php?sClass=version&iId=25350 hmm doesnt sound to bad, lets give it a

https://bugs.freedesktop.org/show_bug.cgi?id=59168 Priority: medium Bug ID: 59168 Assignee: nouveau at lists.freedesktop.org Summary: [nvc3/Quadro2000M] graphic garbage/corruption/noise on resume QA Contact: xorg-team at lists.x.org Severity: major Classification: Unclassified OS: Linux (All)

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PSTN->SIP. We use Cisco 7940 handsets and we also

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Josh Coalson wrote: > --- engdev <engdev@ozemail.com.au> wrote: > >>Hi, >> >>Has there been any progress on the ACM for FLAC >>that was being looked at by Steve Lhomme? > > > Haven't heard anything here about it for a long time. Sorry, I sent my reply to engdev privately (because there's no good reply to). In short I haven't worked on