This month, we're going to perform a basic Asterisk installation and configure two extensions. For this, we'll need the Asterisk@Home distribution, a PC, and two VoIP phones.

Asterisk@Home is a CD-based Linux and Asterisk installation system. Version 2.0 installs Centos 4.2, Asterisk, and a handful of tools such as Asterisk Management Portal, Flash Operator Panel, and SugarCRM. You'll need to download the ISO from http://asteriskathome.sourceforge.net/ and burn it to a CD.

The PC requirements are rather minimal. My development PBX is an AMD Duron 800 with a 40 gigabyte hard drive and 512 megabytes of memory. This low-end system is an ideal test platform and would work fine in a residence or very small office with a few phones.

Speaking of phones, I've selected the Grandstream Budgetone BT-101, an entry-level SIP phone. While these only support one-call presentation ("line"), they work very well with Asterisk, are inexpensive, and have good sound quality from both the handset and speakerphone. The Budgetone can be purchased for $59.95 each from The VoIP Connection, www.thevoipconnection.com/.

Let's get started building our VoIP-enabled, two extension PBX out of a worthless computer, two $60 phones, and a CD downloaded from the Internet.

Installing Asterisk@Home
You've burned the Asterisk@Home CD and the PC has power, a network, a monitor, and a keyboard connected. Asterisk@Home will erase everything on the system's drive, so make sure the machine doesn't have anything important on it.

Put the Asterisk@Home CD into the computer, power it up, and press enter at the boot prompt. This starts the CentOS installation. When the installation is complete, the CD will eject. The computer will reboot and automatically compile Asterisk from source. Feel free to leave the system unattended, as this portion of the installation process is entirely automated, may take some time, and is not particularly entertaining to watch.

When the installation is complete, a login prompt will appear on the console. Type root and press enter. The default password is password.

Once you're logged in, you should see:

Welcome to Asterisk@Home
-------------------------------------------------
For access to the Asterisk@Home web GUI use this URL
http://192.168.1.138

Your URL will probably be different. If your URL is blank, there's a problem with your network adapter or DHCP is not enabled on your LAN. If this is the case, set up DHCP or type netconfig, assign the address manually, and reboot.

While we're logged in, we should change the root password. To do this, type "passwd" and enter the new password twice:
[root@asterisk1 root]# passwd
Changing password for user root.
New UNIX password:
Retype new UNIX password:
passwd: all authentication tokens updated successfully.
[root@asterisk1 root]#

Now that the password has been changed and we know the IP address of the Asterisk machine, log out by typing "logout".

Configuring the Phones
Next, let's set up our two Grandstream phones: plug both into power and ethernet. Once the phones boot up, press "Menu" and use the up/down arrows to cycle through the options until you see "IP Addr". Do this for both phones and note the IP addresses.

If a phone didn't get IP addresses from DHCP, it will set a default address of 192.168.0.160. Pick up the receiver and hang up. This will put the phone back to a ready state.

To set the address:

• Disable DHCP by pressing "Menu." If the phone says "DHCP ON," press the "Menu" button and an up or down arrow to toggle DHCP off.
• Press "Menu" again, then the down arrow. The phone should display "IP Addr".
• To change the address, press "Menu" and enter the new IP address for the phone, using three digits for each portion of the IP address, for example, for 192.168.1.20, enter 192168001020.
• To save the address, press "Menu" and reboot the phone.

The phone will display the Advanced Settings. Enter the settings shown in Figure 1.

Make sure your SIP Server is the IP address of the Asterisk machine. Set the password to "testphone1". We'll also set the SIP User ID and Authenticate ID to 200.

Scroll down and set "Register Expiration" to five minutes as in Figure 2.

Set the Voice Mail UserID to *97 (see Figure 3):

...and "Send DTMF" to RTP/RFC 2833 (see Figure 4).

To save, scroll to the bottom of the page and click "Update" (see Figure 5).

Changes will take effect after the next reboot. You can either cycle power to the phone or click "Reboot" after the configuration is saved.

Repeat these steps for the second phone, making sure to set the User and Authenticate ID to "201" and the password to "testphone2" (see Figure 6).

Configuring Asterisk
Let's take a quick tour of the GUI configuration and administration tools that come with Asterisk@Home (see Figure 7). From another computer, open up a browser and type in the IP address you saw when you logged into the console.

• Voicemail and Recordings: Allows users to check their voicemail through a Web browser.
• CRM: The highly acclaimed SugarCRM Customer Relationship Management system.
• Flash Operator Panel: A powerful Web-based tool that shows the status of every extension, allows an operator to transfer calls, view conference status, view queue status, and more.
• Web MeetMe Control: A Web interface to the conferencing system.
• Asterisk Management Portal: "AMP" is a completely Web-based system for configuring and managing an Asterisk PBX.

To tell Asterisk about our two new phones, click on Setup, then click Extensions. Since our Budgetone phones are SIP devices, select SIP from the Device Technology Menu.

We're setting up the first phone, extension 200 (see Figure 9). The password or "secret" matches the password in the phone of "testphone1." We've set the Display Name to "Test Phone 1" and a voicemail password of 12345. If you'd like Asterisk to send e-mail when you have new messages, enter the e-mail address. To have Asterisk attach the message to the e-mail, enable "e-mail attachment."

When complete, click "Add Extension." A red banner appears as in Figure 10 ...click the banner.

Set up the second phone, using extension 201 and password/secret as testphone2.

Both phones should have a dial tone. If they don't, power cycle them.

Testing
A quick way to confirm proper operation is to use the built-in Echo Test, which plays back everything it hears. To use the echo test, dial *43. You should hear everything you say perfectly clearly, with a fraction of a second of delay.

Fun
Asterisk@Home comes with some useful features out-of-the-box. Table 1 shows some of the most useful.

The weather forecast uses the Festival Text-To-Speech engine, which is installed as part of Asterisk@Home. If you'd like to change the forecast city, I have a guide on my blog: http://steve.linuxworld.com/.

Next month, we'll connect our Asterisk machine to the public telephone network with a voice-over-IP provider, get a toll-free number, and configure the Digital Assistant to answer our incoming calls.

Resources

• Asterisk@Home: http://asteriskathome.sourceforge.net/
• The VoIP Connection: www.thevoipconnection.com/
• Grandstream: www.grandstream.com/
• Author's Blog: http://steve.linuxworld.com/