similar to: My spa has a mind of its own

Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

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

Yes there is - both LiveIce and mpegmixer used a terminal as an interface - telnet/ssh into the box and control it all. Of course back when mpegmixer was released (1996) there was no streaming solutions available so you could only remotely control stuff within earshot ;-) > > But there is no real "remote DJ"-software yet, is there? I mean, not > just "autofades",

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, > >>>

Hello there! our company (e-learning) is trying to roll out Mumble for our users. The quality is awesome but setting up the program is a pain for our users and we'd like to improve that for our users and for Mumble users in general. The main issue our users have is with voice detection. Speaking with Mumble's community they informed me Now, one of the most twitchy aspects of the audio

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

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

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

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