similar to: SIP audio port usage

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

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 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

> -----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 > >

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

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

Hi, How can I retrieve those voicemails using my ip phone? and how will i confiugre it on asterisk? Please help I'm very new in asterisk. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Do?a Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

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

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, I would like to return a combination of named scopes in a named scope : For example, I have a named scope filter and I want to add a named scope eval_filters like Product.eval_filters([''x'',''y'',''z'']) is equivalent to Product.filter(''x'').filter(''y'').filter(''z'') Anybody know how can I

Hi ! I need to create some pdf from my application. I have a table people with a field address and I need to generate a pdf of label of each address ( I think 10 labels per page). The pdf can have around 500 pages. Which library should I use to do it ? I read some information about different library to generate pdf but I don''t know if all library are a good choice for generate

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

Hi, I use ar_mailer to send emails from my application. I want to build a page where the admin case see the queue of mails and the possibility of remove some mail in the queue if he wants. The email in the queue are save in the AR model Email. I have an AR model Sent where i save the email that the user sent. I would like to add to the model Email a column to save the id of the corresponding

Hi, I have a question on the audio data format that your librarie encodes. Must the audio data be in a planar format or interleaved format ? Thanks in advance, Adrien -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170404/2c2c871e/attachment.html>

Hello everyone, After reading the nice blog post about _ExtInt, I was wondering whether operations on i128/i256 and more generally on integer types in LLVM are guaranteed to be constant time or not. For instance, for now, the x86 & aarch64 backend generate constant time code for additions on i256 integers (see https://godbolt.org/z/xMfkqz & https://godbolt.org/z/jbkSpe), but is there

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't