Submitted by admin on Sat, 01/21/2012 - 20:38
Submitted by admin on Sun, 08/07/2011 - 18:53
This install guide was tested using the Redhat Enterprise Linux v5 distribution known as CentOS .
Most of the install procedures for OpenSIPS assume Debian. I prefer working with RHEL which is the reason for this install procedure. This guide should work on Debian with some minor modifications but I haven't tried.
Submitted by admin on Sun, 07/17/2011 - 16:08
Submitted by admin on Thu, 06/02/2011 - 12:29
These are my notes on how I replace TFTP with FTP for Aastra SIP phone configuration files on a Redhat Enterprise Linux server. With the TFTP port exposed to the internet and no IP restrictions for remote phone configuration the directory is wide open. This makes it easy for someone to obtain extension passwords.
Submitted by admin on Mon, 04/18/2011 - 15:02
This guide is a reference for myself on how to create a RedHat Enterprise Linux v6 template for OpenVZ. I am publishing it in case others find it useful. There are other ways and variations on how one can go about this. The method I will present here is the way I do it.
In this case I used the recompiled from source distribution known as Scientific Linux. This guide should work equally well for any distribution including Redhat, CentOS etc. It also works for v5 with a few minor changes or omissions which should be obvious.
Submitted by admin on Wed, 09/08/2010 - 18:36
Bluebox was formerly known as FreePBX v3. It has now been spun off into it's own project. This version of FreePBX supports Freeswitch in addition to Asterisk.
Bluebox is still a work in progress. We felt it was far enough along to create an install guide for those who want to dip their toes in and try out FreeSwitch at the same time. The Bluebox portion of this guide can be used in conjunction with the Asterisk install portion of one of our other guides although we have not tested the combination.
Submitted by admin on Thu, 10/09/2008 - 15:29
External SIP or in other words SIP through firewalls and routers or more accurately SIP traversal through Network Address Translation (NAT) is arguably one of if not the most common problem people face. SIP is notorious for this. This little article is my attempt to provide a simple no nonsense method of finding and fixing these problems. Ideally before they ever happen. To keep it simple I will assume everything on the PBX itself is configured correctly.
Submitted by admin on Tue, 06/03/2008 - 19:42
Here are a few advanced topics covering some things that can be done to any FreePBX or strictly Asterisk distribution. These would be things one might be interested in doing beyond the standard Linux-Asterisk-Freepbx installation.
Submitted by admin on Fri, 04/25/2008 - 14:41
This is a general overview of some of the more common mile high mistakes people make when doing an open source VoIP PBX project. I won't be going into details about hardware dimensioning or software configuration. I will just be providing a general overview of some of the system level common sense mistakes people make in terms of a typical consultant/client type of project.
