similar to: mistake in documentation + a question

Hello! First of all Thanks for such a great codec! I have noticed a few smaller inconsistencies in the documentation, which may be confusing: * Encoder related CTLs OPUS_GET_SAMPLE_RATE(x) "Gets the sampling rate the encoder or decoder was initialized with. This simply returns the Fs value passed to opus_encoder_init() or opus_decoder_init()." ---> Is it a generic CTL?

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

Hi everyone: I use the protel CAD software. In early versions this work well, but in this version i got a little problem Whe execute in terminal appear this into the screen, no other message no fixme, no warning only this fixme:int:WIN87_WinEm87Info (0x7e285aee,12), stub ! But the things that must appear at the botom of window have a extremly big bottom border. so big that i cant see the

The Wine development release 1.1.4 is now available. What's new in this release (see below for details): - Substantial chunks of WinHTTP are implemented. - More JavaScript support. - Beginnings of shell AppBar implementation. - Several fixes for Google Chrome support. - Chinese translations. - Various bug fixes. The source is available from the following locations:

Opus 1.0.2 fixes an out-of-bounds read that could be triggered by a malicious Opus packet by causing an integer wrap-around in the padding code. Considering that the packet would have to be at least 16 MB in size and that no out-of-bounds write is possible, the severity is very low. This new release also has the following changes: Quality-impacting - Changed the behaviour of the PLC to always

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

> In particular, (I'm not really sure, because I don't thorougly > understand it yet) I don't think your jitterbuffer handles: > > DTX: discontinuous transmission. That is dealt with by the codec, at least for Speex. When it stops receiving packets, it already knows whether it's in DTX/CNG mode. > clock skew: (see discussion, though) Clock skew is one of the main

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

I've been scratching my head for a while and I expect it is my mediocre knowledge of Asterisk which is holding me back. If anyone can assist me with some pointers I'd be grateful. Basically, I've hooked up a Viking intercom at the front door. It hooks into an fxs as a "phone". Up till now I've just played back a "go away" message if any internal phones are

> > 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, The attached patch fixes an incistency in my earlier patch. Whereas the rest of the AGC ctls are in dB, GET_AGC_GAIN was linear. This patch fixes that. It also changes the API for _GET and _SET_AGC_LEVEL to use a int32 instead of a float, meaning we don't need to do a API change when we get a fixed point AGC. Best regards, Thorvald -------------- next part -------------- ---

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

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

Hi All, I am trying to recompile the stock kernel to include XFS. At install time of the OS I select to install the kernel source code however I do not have a /usr/src/linux-2.6.9-5.0.3.EL-smp directory. I then learned you must run "yum install kernel-sourcecode" to properly install the source. When I run the command this is what happens ( see below) . The error I get is: warning:

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

On 2020/10/30 ??4:47, Michael S. Tsirkin wrote: > On Tue, Oct 27, 2020 at 12:47:34AM -0500, Mike Christie wrote: >> On 10/25/20 10:51 PM, Jason Wang wrote: >>> On 2020/10/22 ??8:34, Mike Christie wrote: >>>> Each vhost-scsi device will need a evt and ctl queue, but the number >>>> of IO queues depends on whatever the user has configured in userspace.

So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I have the following gripes, which I've sent to a very clueful Cisco person already. Mind you, I love the Cisco 79xx series phones, and currently they are what I recommend to anyone who wants a 'real' IP phone. I just cringe - Speed dials. It's nice to now have speed dials in the line appearances that

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has

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,

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?