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I was unable to build the whole Mumble system on CentOS but it's available for Fedora Development. I just wanted the server part on my headless server, and a static build is available from the Mumble project on Sourceforge. So I grabbed the Mumble SRPM from Fedora Development, the static build from Sourceforge, and stripped the spec file down to the minimum needed to just install the

Hi, When compiling mumble-iphone from https://github.com/mumble-voip/mumble-iphoneos <https://github.com/mumble-voip/mumble-iphoneos> I had a parser error on the file CodeOutputStream.m (form https://github.com/booyah/protobuf-objc/tree/696b7b61cdd4e8d77c55ace98b3119194fa04b7f <https://github.com/booyah/protobuf-objc/tree/696b7b61cdd4e8d77c55ace98b3119194fa04b7f>). I know that the

On Fri, 19 Aug 2016 23:40:45 -0700 ToddAndMargo via samba <samba at lists.samba.org> wrote: > On 08/16/2016 12:52 AM, ToddAndMargo via samba wrote: > > Hi All, > > > > Fedora Core 24, x64 > > samba-4.4.5-1.fc24.x86_64 > > > > smbclient -L 127.0.0.1 > > > > is very, very slow. It comes back correctly, but takes 20 seconds. > > One

somefile.pp class mumble { } class mumble::testcontents inherits mumble { file { "/tmp/testfile": path => "/tmp/testfile, ensure => present, mode => 0640, content => "I am a test file.", puppet parser validate passed it. puppet apply says it is run yet no /tmp/testfile appears the puppet master is running and this is being

On Sat, 20 Aug 2016 04:21:07 -0700 ToddAndMargo via samba <samba at lists.samba.org> wrote: > On 08/20/2016 01:35 AM, Rowland Penny via samba wrote: > > On Fri, 19 Aug 2016 23:40:45 -0700 > > ToddAndMargo via samba <samba at lists.samba.org> wrote: > > > >> On 08/16/2016 12:52 AM, ToddAndMargo via samba wrote: > >>> Hi All, > >>>

I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1

Does there exist *anywhere* a Java web browser plugin for 64-bit FireFox? The SUN 1.6<mumble> JDK (jdk-6u18-linux-amd64.rpm) does NOT include the Java web browser plugin library. java-1.6.0-openjdk-1.6.0.0-1.2.b09.el5.x86_64.rpm does not have one either. Should I install the *32-bit* SUN 1.6<mumble> JDK and use the 32-64 bit wrapper? I've searched the web and read the wiki

Hallo I'm new in the mailing list :) I'm working with a program called mumble that uses speex and the problem is that, if I use a mumble client that tries to reproduce a wave file containing a 1Khz tone, I cannot hear that tone. I've tried to comment all the denoise functions but nothing could solve the problem. I've turned off the VBR, with the function speex_encoder_ctl and

I'm seeing grep break when I grep large numbers of gzip files in /var/log. Posted bug on bugzilla. https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=205534 Description of problem: grep errors with: --- *** glibc detected *** free(): invalid next size (normal): *** Aborted. --- when doing recursive grep in /var/log, other places. Version-Release number of selected component (if

For some awful reason (can someone explain why?), the RHEL/CentOS build of firefox is hardwired to use evolution as its mailto client. With FF 2.<mumble> this was fixable with an about:config setting. With 3.0.<mumble>, it seems not. What is the proper way of fixing this? *I* ended up doing a 'sudo rpm -e evolution' and then doing a 'sudo ln -s

> The underlying liveness datastructure is a list of ranges where each vreg is alive > (ranges in terms of instructions numbered). I remember a couple of later linear scan > papers describing the same thing (Traub et.al. being the first if I remember correctly). > That should be as accurate as you can get in terms of liveness information. It depends on the details. For example, given

Thanks Dan - might have to scratch my head over that one for a while! The phrase "you make your own RTP server" has made me all twitchy ;) Jonathan On Wed, 6 May 2020 at 07:21, Dan Jenkins <dan at nimblea.pe> wrote: > Hi Jonathan, > > I'd probably go down the external media route in the ARI now - you make > your own RTP server and provide your own RTP back to

Use apps like mumble of Teamtalk, they are build to do what you want to do. Kind regards, Dennis Op 19-9-2015 om 00:32 schreef Marvin Scholz: > On 18 Sep 2015, at 23:31, Orion Jensen wrote: > >> I'm investigating the build out of a Push to Talk server with multiple >> subscribers as part of a mobile app. > That most likely will not work with Icecast, you might want to

> When enabling and compiling this with mingw on Win32, my debugger > complains about heap overruns when calling speex_encoder_destroy(). This > could be a mingw issue though, as I also found a stackalignment bug which > prevents me from _USING_SSE (apparantly -mpreferred-stack-size is just > "prefered" and therefore ignored... *Sigh*). Just so I understand, the

The PGO was my first guess but I can get a lot of information. At first, I follow the explanation at http://clang.llvm.org/docs/UsersManual.html#profiling-with-instrumentation but instead of llvm-profdata merge, I used llvm-profdata show *.profraw. Sadly, the information I get is the total number of function, the maximum function count and the maximum internal block count. Do you know if you

The phi instruction is irrelevant; just the way I think about things. The question is if the allocator believes that t0 and t2 interfere. Perhaps the coalescing example was too simple. In the general case, we can't coalesce without a notion of interference. My worry is that looking at interference by ranges of instruction numbers leads to inaccuracies when a range is introduced by a copy.

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my

Hello Campers I''m not sure this is the right list to ask but: I''m using the junebug wiki (which is written in camping) for my own personal wiki. I''d like it to behave as a proper daemon, so it will start up at boot, get stopped properly at shutdown, keep a pid file in /var/run etc. This requires a little messing with the wiki script junebug makes in the wiki

Using ALSA or OSS instead of PulseAudio is not actual at all. Its si much problems using ALSA and OSS with duplex sound thru wine and native linux-apllications. And problems comes when trying to use things like mumble and playing a game with wine. The pulseaudio-patch which was added for Wine versions up to 1.3.28 (not official i think), has been working just perfect. I tried 1.3.34 which

Is there an official speex logo anywhere? As I owe the voice quality in Mumble to Speex, I'd very much like to add at least an "About Speex" dialog, but it will look kind of dull without at least a tiny icon.