similar to: Important :: Please support the development of a new Jitterbuffer for SIP

Hi, Did someone ever managed to make Astertest (http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed all the instructions of this tutorial and corrected the mistakes pointed by the users but it still doesn't work. I can compile it and load app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I have this error : WARNING[31477] : loader.c: 325

Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To

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

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider

In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ /O

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.

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

It there avalible quality and delay test for sip connections for asterisk. Something like to clients making a call with different codecs and measuring delay , jitter ? I know there is a Astertest but in that you need 2 asterisk mashines (which is usually hard to have). I was looking for perl/bash scripts running sip clients in a finite loop + etheral to measure packet properties , gathering logs.

We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us

I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa

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

Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is

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

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

Hi all!! I've made some changes to the applications that Astertest was using to monitor the performance of the server. Now is also possible to track the bandwidth usage of the server, this has nothing to do with the executable (astertest.exe) itself but with the events that the Asterisk Manager generates. The method described in: http://www.asteriskguru.com/tutorials/astertest.html to

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

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,

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