similar to: kernel-rt on CentOS 8

Hello everyone, Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems with high loads (and probably any system at that) - especially if it is finely tuned and has zaptel

Hello everyone, (I first sent this several hours ago, it appears it got lost. Thought I'd give it a second try). Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems

Hi, I am writing because I would like to know if centos will update the kernel-rt package from https://git.centos.org/rpms/kernel-rt/releases ? I ask because bz#1550584 https://bugzilla.redhat.com/show_bug.cgi?id=1550584 mentions "fixed in version kernel-rt-3.10.0-1063.rt56.1023.el7" And the latest tag in "releases" link above is for kernel-rt-3.10.0-1062.7.1.rt56.1030.el7

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

Hi All, I need to build clang/llvm from source on an Ubuntu Linux system, in order to build the gnustep Objective-C environment. I was able to build clang using cmake without a problem (following the instructions at clang.llvm.org/get_started.html), but when I tried to build gnustep, I found that clang failed to compile an autoconf-generated program. The compile error in the configure log was:

Building the BlocksRuntime isn’t supported via CMake at the moment. It should be pretty straightforward to make it work, but it currently isn’t supported. -Chris > On Sep 29, 2015, at 9:45 AM, Alex Wang via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > As far as I know the makefile in there isn't the makefile that gets executed during the cmake build -- cmake generates its

Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.com =

> On Jan 30, 2014, at 1:50 PM, Reid Kleckner <rnk at google.com> wrote: > >> On Thu, Jan 30, 2014 at 1:19 PM, Renato Golin <renato.golin at linaro.org> wrote: >>> On 30 January 2014 20:33, Reid Kleckner <rnk at google.com> wrote: >>> Basically, compiler-rt contains the only runtime libraries we ship with Clang. The sanitizers are runtime libraries

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

Has anyone tried to use Wine with realtime Linux? I am considering porting a system which currently runs under "realtime" NT to RT-Linux. The non-realtime portion is a GUI front end written with Borland C++ Builder, and I would prefer not to have to rewrite it. Is it feasible to run Linux as a low priority task under RT-Linux and to then run the windows app with Wine? Would the

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

Hi, I searched the Internet to find an answers, but I didn't find anything useful, so I'm turning to you, maybe someone has the answer. I want to make a realtime broadcast from a Linux box. The source is the soundcard's line-in, and it sends the stream to an Icecast server. I would like to have realtime compressor/limiter functionalities on this Linux box, so the outgoing signal

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

builtins/clear_cache.c depends on kernel headers. It's difficult to ascertain whether it's mixing with libc headers but if it is then that's a bug in its own right. Either way, it's pulling in kernel headers for a single #defined constant that's been part of the linux ABI for years now. This makes it more difficult to cross compile for, in my opinion, no good reason. So

For me.... The latest RT kernel, 2.6.31-3-rt works for wine incoming sound, provided you killall pulseaudio first. The application I tried it with was Dragon NaturallySpeaking. It works when setting either alsa or oss in winecfg. However, the generic kernel, 2.6.31-9-generic, still shows a timeout error. Conclusion? Incoming sound in wine -- Maybe use the RT kernel if you have trouble getting

hm, to have a look at Pure Data and external called oggcast~ http://www.akustische-kunst.org/puredata/ is recomended, you can set quality/bitrate/samplerate ... while streaming then, and do whateweryou want to your sound before sending it to icecast, including building your compressor-limiter. cheers Ales Zemene -- http://ales.mur.at irc.kunstlabor.at #kunstlabor citation of

Hi, I think this more of a front-end issue than a compile-rt issue, but I’m also copying the llvm-dev list In compiler-rt the file lib/builtins/atomic.c seems to rely on determining at compile time if an atomic operation of size 1, 2, 4, or 8 is always lock free. For example, in the implementation of __atomic_load_8() we have something like this after macro expansion: void __atomic_load_8(…) {

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Ubuntu 10.04. Wine 1.2 I'm running Native Instruments Kontakt 4 under Wine. It runs fine except for occasional stutters, so I wanted to try enabling Realtime in Jack, but I have no luck with this. (I have made recommended edits to the limits.conf file, though I've read that this is no longer necessary(?): @audio - rtprio 100, @audio - memlock 250000) When I start up Kontakt, I see