similar to: help sip show on CLI : no such command

On May 3, 2006, at 9:54 PM, Jean-Marc Valin wrote: >> Perhaps, but then you need to assume that the jitterbuffer can just >> throw away the data, and that limits how you can use it. In object- >> oriented terms, you might want to pass objects to the JB, and then >> call a destructor on them. In C terms, you may want to allocate >> frames via malloc(), and then call

Hi guys, i've been using iax2 show netstats and i wonder if someone could explain what all these means, just in case i have them wrong. Because i am looking for something that tells me that there is delay , and/or packet loss. -------- LOCAL --------------------- -------- REMOTE -------------------- Channel RTT Jit Del Lost % Drop OOO Kpkts

I wonder if there is a GUI that does not change the underlying hand-made configuration ?! What I'm looking for actually is a GUI for adding a new SIP-client + voicemail, so that a company does not have to call me when they hired a new employee. I don't want a GUI that over-writes my hand-made SIP-configuration, and my hand-made dialplan. Jonas. -------------- next part -------------- An

> > 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 tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

On May 3, 2006, at 9:12 PM, Jean-Marc Valin wrote: >> We just return a frame with the return value JB_DROP, which tells the >> caller to drop this frame, and call jb_get again. >> >> When the caller is done with the jitterbuffer, it calls jb_getall() >> repeatedly, until it's empty, and then it can discard all the frames. > > Hmm, looks a bit error-prone to

On May 3, 2006, at 7:40 PM, Jean-Marc Valin wrote: > >> Yes. Jean-Marc has made the API more similar. >> >> Jean-Marc: Have you looked at the API we have for the >> asterisk/iaxclient jitterbuffer? > > Just did. > >> It's pretty close to what you have now -- the major difference is >> that >> your jb still assumes it can

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

Jean-Marc Valin wrote: >>OK, I'm actually about ready to start working on this now. >> >>If people in the speex community are interested in working with me on >>this, I can probably start with the speex buffer, but I imagine >>there's going to be a lot more work needed to get this where I'd like >>it to go. >> >> > >And where

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.

> Perhaps, but then you need to assume that the jitterbuffer can just > throw away the data, and that limits how you can use it. In object- > oriented terms, you might want to pass objects to the JB, and then > call a destructor on them. In C terms, you may want to allocate > frames via malloc(), and then call free() on them later. You might > want to pass in

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <div class="moz-text-flowed" style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi, <br> <br> I try to find a solution for this problem : <br>

Mike Taht wrote: > > > On 5/3/06, *Jean-Marc Valin* <Jean-Marc.Valin@usherbrooke.ca > <mailto:Jean-Marc.Valin@usherbrooke.ca>> wrote: > > > I must say I really like the generalized jitter buffer though :) > It's a > > cleaner and more flexible implementation and can more easily be > adjusted > > to contain additional

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

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

I know, that jitterbuffer should be set at receiving side and on outgoing call leg, ie. if sipphone calls to asterisk and outgoing to zap chanel, I should set jitterbuffer on zap channel (to dejjitter audio stream from sipphone) but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)? I have following setup (homeworkers using sip phone connected to home asterisk via SIP and

> Yes. Jean-Marc has made the API more similar. > > Jean-Marc: Have you looked at the API we have for the > asterisk/iaxclient jitterbuffer? Just did. > It's pretty close to what you have now -- the major difference is that > your jb still assumes it can "own" the data passed in -- it copies it, > and it destroys it at will. With the API I put together,

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?