search for: thomsons

...let your Android phone connect via LTE > to your home Asterisk server and you dial your (home, cabled) Thomson phone > from it? What's the call quality like then? The quality is terrible. It is not possible to understand any word... BUT: if I call my wife using the Thomson (she uses a Thomsons, same model, too!) the quality is excellent... > In regard to: > > On Saturday 13 June 2020 at 18:25:32, Luca Bertoncello wrote: > >> 2) where can I change these settings? > > sip.conf > > Look for lines such as > > disallow=all > allow=ulaw > all...

Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to

Hi, I''m fairly new to Rails and v. new to RSpec. Think it looks pretty useful and so I''m trying it out for the first time. I have installed the rspec-rails gem and created a simple plain text my_story file and my_story.rb file along the lines described here - http://www.tomtenthij.co.uk/2008/1/25/rspec-plain-text-story-runner-on-a-fresh-rails-app. When I run

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the

Taken from http://www.twice.com/html/pagebeta.cfm?InputKey=2853 ------------------------------------------------------------------------- Thomson, Fraunhofer Developing Enhanced MP3 Codec Called mp3PRO Jan. 7, 2001 By Joseph Palenchar Las Vegas -- Thomson and the Fraunhofer Institute, the co-developers of MP3, are working with a third company to develop an improved MP3 codec intended to deliver

On Saturday 13 June 2020 at 22:30:28, Luca Bertoncello wrote: > 1) I have an Android phone, using the integrated Android VoIP-subsystem, > connected to my Asterisk at home, over LTE or other network *outside my > home network*. > I called my mother using this method... The quality was excellent > 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet > (cabled

Thanks, John. This sounds reasonable that the analysis group just provides an interface to its implementors (pass). Then I have another question, is it allowed to specify multiple pass in the same analysis group in the command line for opt? If yes, which pass will be used for the call of getAnaysis<AnalysisGroup>()? -Thomson On Mon, Jun 25, 2012 at 10:34 PM, John Criswell <criswell at

I found that LLVM optimized the IR by replacing printf with puts. I wondered which pass did this optimization? And is it common that puts is faster (and some other metric) than printf? -- Thanks Thomson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20120617/08aa6c45/attachment.html>

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.

On 6/25/12 8:55 PM, Thomson wrote: > Thanks, John. > > This sounds reasonable that the analysis group just provides an > interface to its implementors (pass). Then I have another question, is > it allowed to specify multiple pass in the same analysis group in the > command line for opt? Yes, you can do that. Using multiple analysis passes that belong to the same group may

Suppose we have a data file containing: "Smith, John", 120, 90 "Thomson, Peter", 110, 85 there are 3 variables in it. If we use x <- read.table("tmp.txt", sep=",") to read the data to a data.frame, the result will be 4 columns. Splus 4.0 have no problem with this kind of data. Li Dongfeng

Suppose we have a data file containing: "Smith, John", 120, 90 "Thomson, Peter", 110, 85 there are 3 variables in it. If we use x <- read.table("tmp.txt", sep=",") to read the data to a data.frame, the result will be 4 columns. Splus 4.0 have no problem with this kind of data. Li Dongfeng

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

I have tried 4 time unsuccessfully to install Xen 4 on Ubuntu Lucid (10.4) 64 bit version using the tutorial at http://bderzhavets.wordpress.com/2010/04/24/set-up-ubuntu-10-04-server-pv-domu-at-xen-4-0-dom0-pvops-2-6-32-10-kernel-dom0-on-top-of-ubuntu-10-04-server/ on a Lenovo thinkcentre workstation. When I boot I get a kernel panic. If I boot the Linux kernel that I compiled using the

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

Hi Thomson, the new call to puts() is inserted right away, whereas the old call to printf() is removed a bit later in SimplifyLibCalls::runOnFunction(). If you browse the code a bit and backtrack the call stack to see what happens with the return value of PrintFOpt::OptimizeFixedFormatString(), you will stumble upon this segment in SimplifyLibCalls.cpp:1703ff.: // Try to optimize this call.

I found that ProfileInfo is contained in an analysis group, but ProfileInfo class itself is not derived from any base classes of passes, so it appears not to be a pass. Anything I missed here? Thanks, -Thomson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20120624/7f26ca93/attachment.html>