Last day I was figuring out Asterisk (an opensource pbx solution) I have a lot of experience with CME and a little with CCM from Cisco Systems, but I believe companies can save tons of money with an opensource solutions for their VOIP solutions.
I used Trixbox version trixbox-2.8.0.4
Kernel Version 2.6.18-164.11.1.el5 (SMP)
Distro Name CentOS release 5.5 (Final)
This is the current stable release is based on Asterisk 1.6
And really it’s just an asterisk with extra tools onboard. You can even install clean Linux then Asterisk then FreePBX. But I like to have an all-in-one installation.
I first used VMware player for this but it was so slow I gave it up.
For further investigation, I used an intel 2.6dualcore, 1G mem. 200GB HDD 7200rpm setup.
So after the installation, log in with the username: root and the password you made during installation. type: ‘system-config-network’ pick your network card and set a static IP address/default gateway. Save and type in the console ‘service network restart’
Note: type in the console ‘asterisk -r’ to access the asterisk console.
Go to ip you just provided through Firefox (IE is buggy)
Click on Switch next to User Mode this will open the Admin GUI. Login with
user: maint
pass: password
First, we install the DNS really quickly, go to System> network> Edit network parameters (hit Use OpenDNS)
And created an extention> PBX> settings> extensions >add extension > choose for SIP Device
This is what I filled in, leave anything else blank:
User Extension 7111
Display name:Andy Stevens
Secret 7111
I used a softphone for testing (x-lite)
Here are the settings:
Display Name: Andy Stevens
User name: 7111
password: 7111
Domain: <IP you just created)
Register with domain and receive incoming calls (selected)
Target domain (selected)
Next register a user account at voipcheap.com (download their client for doing this)
Then we make a trunk: PBX>settings>Trunks>Add trunk
Here is the config leave everything else blank:
Maximum channels: 2
PEER Details:
allow=alaw&ulaw&G.726&gsm&g729&g723&g777
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromdomain=sip.voipcheap.com
fromuser=username
host=sip.voipcheap.com
insecure=very
nat=YES
port=5060
qualify=yes
secret=password
srvlookup=yes
type=peer (VERY IMPORTANT FOR 2 way CALLS!)
username=username
User Context:: leave blank
User Details: clear dummy stuff
Register String: username:password@sip.voipcheap.com/username
Then go to Outbound routes> add route:
Dial pattern: 9| xxxxxxxxxx (just for the test we let outside calls happen when pressing a ‘9’ < 9 is striped off)
Trunk Sequence: Select your trunk u just created.
Then go to inbound routes and set the destination to the 7111 extension
A working senario :)))
Cheers!