similar to: jitterbuffer in pure voip (sip/iax) - what is best practice

my biggest concern about defaulting the context to anything at all besides [default] is that you then have to remember to configure the voicemail.conf with the corresponding contexts. as it stands, you have the ability to do just that, but you don't have to. if you have several hundred extensions broken out by dozens of contexts, it might not make sense to force the voicemail.conf to follow

http://bugs.freedesktop.org/show_bug.cgi?id=24596 Summary: Video adapter #0 is jittery, while adapter #2 isn't. Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at

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

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and

Hello all, We've been using our Asterisk system live for about a month now and I'm looking to tuning a few things. First, is echo, I receive a fair amount of echo during the first 10-15 seconds of incoming calls. Next is a very weird problem. We have serveral Polycom IP300's and one Budgetone phone. It seems that if we unplug & move the Budgetone (which happens a fair amount

On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > the JITTERBUFFER function?

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

Hi, I have a question about jitterbuffer and PLC. I use Asterisk 1.6.2.0 and 1.6.0.20 or older. I use uLaw. My system map: ============================================================================= [ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ] ============================================================================= I use two asterisk server.

Immediately after running winetricks a month ago, several very important directories disappeared from my hard drive, and despite paying a lot of money to good data recovery company, I have not been able to recover them. I have searched this forum and elsewhere for evidence of this happening to anyone else with no result. Is it at all possible that the script deleted my directories by accident

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having

WTF is a jitterbuffer? Sent from my Verizon Wireless 4G LTE smartphone -------- Original message -------- From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM,

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

Greetings, I have a Sipura SPA-1001. When I make outgoing calls, I have very jittery sound. Incoming calls work fine. This wasn't the case a few months ago, I am running head as of yesterday. Any suggestions? Thanks, Erik

hello, is there any equivalent, that is currently usefull, if I have some iax connections with jitter spikes and another with minimal jitter? for my jittery connections, I don't like to shrink jitter buffer too fast, because another jitter spike can occur again and small jb can't cover it. as I read, in older iax jb implementation, this can be solved using "jittershrinkrate="

Hello there, I have set a simple environment to test some functionalities of asterisk's new jitterbuffer. The environment is composed of a sip softphone registering in asterisk 1.4 and calling a pstn phone connected to asterisk through a fxs board. Using the fixed buffer implementation the call quality is improved when injecting a artificial jitter in my local network. However, when changing

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share