10-minute guide to Asterisk

• Download the tarball: Download the Asterisk stable distribution tarball from http://www.voip-info.org/wiki-Asterisk-mirrors.

• Build Asterisk: Unpack the tarball and run make to build it. After that, run make install.

• Install sample configuration: Run make samples after make install. This will install the standard sample configuration in the directory /etc/asterisk. Go there.

• Start Asterisk: The first time, start Asterisk in console mode with some debugging applied. If it does not start, check the hardware requirements.

• Don't change anything yet: Try to start Asterisk with NO changes to the config files. If it doesn't start then the problem is probably with your system hardware.

• Create a SIP account: Copy one of the device configs in the SIP configuration file sip.conf and change for one of your phones. You should only have to change the device id. Don't get fancy with authentication or NAT. Make sure the phone (see Asterisk phones) and your Asterisk server is on the same network, with no firewall and NAT device between them. If you want to learn more on how to get this configured, read the introdcution articles referenced to on the Asterisk page, the bottom half. (If you have a physical card, you can configure a card like an X100P instead of configuring SIP by making 2 small changes, see: http://www.digium.com/downloads/hw_article)

• Restart Asterisk: Either issue the RELOAD command at the command line interface or completly shutdown and restart Asterisk. (RELOAD should be sufficient but one of the 2 is required)

• Call the demo: test that phone by calling extension 1000 (assuming you have the sample extensions.conf). You should get the demo greeting and be able to do such things as ECHO test, leave voicemail for a sample mailbox. Instructions are in the demo greeting you will hear. How this works is explained in the extensions.conf demo file installed for you in the /etc/asterisk directory.

• Add another phone: If all of the above works then add your 2nd phone into sip.conf and add a dialplan for those 2 devices to extensions.conf so that they can call each other. (don't forget to RELOAD or restart)

• Choose a voice: Once things are working, you should decide if you want to use the default voice distributed with Asterisk, or install a professional third-party sound file collection made to replace the default voice. Voice Vector Media and others offer complete replacement sound file collections for free. In addition, inexpensive custom recording services are typically offered to allow you to customize the collection for your needs using the same voice.

• Start exploring Asterisk: If all of that works then start making fancy config files and using additional features. Maybe add a hardware card to connect your PSTN connection to your Asterisk or start experimenting with free SIP accounts on the Internet, like Free World Dialup.

• Asterisk: The start page | Tips & Tricks | Introduction | Applications | CLI