Right, been meaning to play around with this for ages now, just another one of them little projects which iv never got a spare machine lying around for.

Asterisk, for all those who don’t have a clue, is a Open PBX platform for Linux, allowing users to create their own home, or business phone system (Great Fun). Iv been running my VoIP number though a friends server for the past year, whic worked fine while down in the deep south, alas Virgin Media and their wonderful internet connection dont seem to have very good links between the midlands and london (big surprise), as a result, phone calls sounded like i was talking to the HAL-9000 while he was having his balls cut off !

This was my chance, Asterisk here i come… thanks in no small part to the Aussievoip wiki page for documentation and guides, I managed to get an Asterisk & FreePBX system up and running in a day. (it should be noted that there are much easier routes to be taken than the one i took, asterisk now has a version called trixbox and AsteriskNOW, which are full Linux ISO disks which install everything for you, i just prefer to know what its doing).

The FreePBX platform hasa multitude of additional modules which can be used to enhance its uses, including Music On Hold, Calling Queues, etc… which makes it a perfect (Free) application for home users and businesses, sorry BT but someone makes cheaper ones than you lol

Due to some small issues with NAT (our helpful ‘lack of public IP’s’ solution) i had to add a few lines to my sip.conf file, which now reads;

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1″ to each peer definition to
; solve translation problems.

[general]

bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying “The number you have dialed is not in service. Please check the
; number and try again.”

;context = from-sip-external ; Send unknown SIP callers to this context
context = from-trunk
callerid = Unknown
tos=0×68

externip=Dynamic.dnsalias.com
externrefresh=60
localnet=192.168.8.0/255.255.255.0
nat=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf
Basically, making sure that the signaling between my PBX and the outside world was working right, a few extra lines were needed to make sure it could resolve the path back to me ! the problem i was having before this was quite a-typical of NAT issues, callers who dialed in were not audable, they could hear everything i was saying, just nothing was coming from their end, also after 20 seconds Asterisk decides that due to lack of messages from the outside world that this was a spoof, and gave up (nice eh). Problem Solved now… yay

Right, now its all working perfectly, time to break it lol ! nah, im gonna keep running my own box, but im using my old exchange server atm as a test machine for this, so for the full install im gonna be using;

Wall Mountable Micro ATX Case

VIA EPIA ME6000 Fanless Motherboard

2GB IDE Flash Module

All in all, a silent, nicely powerful PBX server for under £150 !

Stay tuned for updates on cheap sip carriers (currently using VoIP cheap, but looking for better !)

——–

P.S just to keep informed, the current version of FreePBX is 2.2.2, version 3 is rumoured to be feature packed and coming shortly to a Linux machine near u !