similar to: Discover Asterisk 1.4 :: Jitterbug, no, Jitterbuffers

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme

Hello, We have chosen to upgrade our Trixbox installations (2.6.2.3, asterisk 1.4.20) and everything work smooth. The problem we face now is that asterisk14-addons-mysql looks to have not been compiled with uniqueID feature and we are asking your opinion about what should be the best fix for this problem. Our workarround was to overwrite (from backup) the cdr_addon_mysql.so module, but this is

It seems that asterisk-addons and one or more of Digium's licensed modules such as res_fax_digium have a conflict that doesn't seem to be documented anywhere I can find. In a nutshell, asterisk14-addons-core has a fake provide for asterisk-gplonly : # # core subpackage # %package core Summary: Asterisk-addons core package. Group: Utilities/System Provides: asterisk-gplonly Provides:

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading?

Jitterbug is just what I want. Bugzilla is great but not for a non-programming (issue-tracking) audience. Can I get jitterbug somewhere please (it seems you have removed it) ? Cheers andy -- andy _______________________________________________ Andy Heath a.k.heath@shu.ac.uk

Hi folks! I planned to put asterisk-1.4 on my slug (one of these embedded devices). Since i don't want to compile it natively on this slow processor, i need a cross-compile toolchain. I tried out several ways such as manually compiling binutils and gcc, or using scratchbox. All of these with the result that i got binaries for machines except the one i wanted! :( I then have been given a link

This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress.

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

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? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

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,

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

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in

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

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

I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with "dead air" or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached

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

I've written a new article about Asterisk 1.4's Jabber integration. Check it out at http://www.voip-forum.com/asterisk/2008-01/xmpp/ /Olle

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.