similar to: Controlling asterisk in a dynamic way

Thank you for your advise, I'll find something with a lot of memory.... Adrien -- Adrien Laurent - CIO 514-284-2020 adrien@modulis.ca www.modulis.ca -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Walker Sent: Saturday, October 01, 2005 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial

Thank you very much Dirk! Le mer. 13 mai 2020 ? 14:59, Dirk Eddelbuettel <edd at debian.org> a ?crit : > > Salut Adrien, > > It appears to be a bad OpenMP and and OpenBLAS interaction you can (for > now) > avoid) by replacing the 'pthread' variant of OpenBLAS with the OpenMP > version > (see the thread for details). Doing > > sudo apt install

Hi Montrealers ! I would like to create a usergroup for Montreal's asterisk users. If you are interested, contact me and we'll schedule a beer/coffee meeting downtown next week. Sincerely, Adrien -- Adrien Laurent - CIO 514-284-2020 ext 202 adrien@modulis.ca www.modulis.ca

Hi, This is a reminder about our next meeting. It will be held from 6pm to 8pm, February 21 at Modulis offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal. Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer. If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to

Hello everyone, I have an IAX server (asterisk@home) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Salut Adrien, On 13 May 2020 at 14:19, Adrien FABRE wrote: | I have upgraded R (from 3.6 to 4.0) and RStudio (from 1.1 to 1.2.5) a few | days ago, and Ubuntu from 18.04 to 20.04 yesterday. | | Since then, R sometimes never terminates when executing certain commands: | ivreg (from package AER), summary (of a logit regression) and logitmfx | (from package mfx). Sometimes these commands run fine,

Hi, I have an IAX server (asterisk@home) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks a lot,

Hi Everyone, I have an old VB6 application that is running fine with wine under X. The application run on command line but use windows if there is an error (which I don't need when running in production) I need to run this application from command line only (I'm calling it from a php script), but MSVBVM60.DLL failed to initiate, when there is no DISPLAY running. This is the error I get

Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

Hi everyone, I'm looking to buy a server that could handle 100 IAX users (g711)-(about 300 registrations) simultaneously. No zap channels. My budget is 1000$ us, Is a fast (3ghz) single server more reliable than a double cpu (like 1ghz) ? Will asterisk take full profit of two cpus? Isn't better to get a second cpu to handle system processes (like stat generation, backups...)? so that the

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

hello.... I'm new to wine and tested and installed it, because I wanted to run the hotline server from windows under linux... upload, news works great but download from this maschine is not working! strange thing does anybody has the hotline server runing? do you have had similar problems with another network app? this are the errors.... imted from C:\windows\system\wininet.dll, setting

The Linux Journal's 2002 Readers' Choice poll is open. Vorbis is featured in the Favorite Audio Tool category, so go vote for your favorite codec! Actually, you even have the choice of voting for "ogg123" or for "Vorbis Tools", since it seems someone at Linux Journal is not quite aware of what Vorbis Tools contain. I suggest voting for "Vorbis Tools", so that

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

I have upgraded R (from 3.6 to 4.0) and RStudio (from 1.1 to 1.2.5) a few days ago, and Ubuntu from 18.04 to 20.04 yesterday. Since then, R sometimes never terminates when executing certain commands: ivreg (from package AER), summary (of a logit regression) and logitmfx (from package mfx). Sometimes these commands run fine, but most of the time I have to kill the process because R won't

Hi all, This is my first experiencing with writing to English-speaking mailing list, so, please excuse me for possible mistakes. I'm a newbie on freebsd. I have a mail server on FreeBSD 4.7 with Postfix and Amavis ( mail virus scanner ). The problem is I got this in my log periodically : /kernel: pid 24499 (unrar), uid 1010: exited on signal 11 (core dumped) I use unrar for Amavis

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Adrien Laurent > Sent: 02 August 2005 14:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to create a secret code to use > myasterisk@home server's long distance plan from a public phone > >

I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger