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. 

Includes every detail in the form of step by step instructions from bare metal to a running VoIP PBX in about 2 hours.  Because it does not require MySql or Apache, Asterisk GUI v2.x is a good alternative to FreePBX for someone looking for a less resource intensive install, possibly on a flash drive and/or embedded appliance hardware with limited CPU/RAM.  The whole install including source files takes up about 1.2Gigs of drive space and uses about 100MB of RAM while idle.  It could probably be reduced to 1Gig of drive space or less with very little effort.  Assuming you

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.

In previous posts I showed how to auto-provision and configure Aastra phones by manually installing XML scripts.  Aastra has released RPM packages for Trixbox that combines everything and adds new features as well as mDNS so the phones can find the TFTP server even if you don't have a DHCP server capable of assigning it.  The script won't

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.