If you are interested or just curious in asterisk you can use the following links to tinker with it.
Asterisk: The Future of Telephony This book is published by O'Reilly and is available from Borders and Barnes & Nobles. The above link is to the free online publication of the Book. This book is a required must-read for anyone starting with asterisk. The current version is somewhat dated and they are in the process of releasing a new edition. This book will apply through Asterisk v.1.2.
VoIP-Info Once you have read through the above book, and are ready for real-world examples and recipes for disaster and enlightenment, this site will be it. It contains a vast array of user-contributed programming examples, command definitions, and other misc topics. It is also a good spring-board to a wealth of information on other sites.
Digium: Creators of Asterisk This is where Asterisk can be freely downloaded and all of the components it needs to run. For POTS you will need Asterisk, Zaptel, Asterisk-Sounds, and Asterisk-Addons. If you are going to experiment with PRI T1's, then you will need LIB-PRI as well. I usually download all 5 and install just as good measure. What's 2-3 megs of wasted hard-drive space
A simple recipe for an R&D Test Bed system would be as follows:
-Pentium MMX 200Mhz or Better (A 1gig P4 will handle 2-T1's with ease)
-Atleast 128-megs of ram
-8-Gig's of Hard-Drive or more
-an X100p card available on E-Bay or
www.x100p.com (They are typically $10-30 depending on where you buy one)
Your favorite Linux distribution (I like Slack

)
This test-bed will let you tinker and play with asterisk for under $200 (at the most). This will let you install, understand, and conceptualize your own dial-plans. One gotcha that this system will have is it will max out around 4-5 lines and the X100p style of cards are horrible, but cheap. Expect to have Gain and Echo issues using X100p style cards. If you want to try a quality interface card for POTS I would recommend Sangoma A200. It comes with a free software echocan for up to 6 channels, or is available with a hardware echocan. T1 Interfaces are available from Digium and Sangoma in multiple-interfaces up to 8-T1's and with or without echocan. Sangoma is also in the works on finalizing a T3 interface. If you plan to experiment solely with SIP/VoIP then just get an X100p as it is used internally by asterisk as a timing source for some functions.
If you want a SIP phone to experiment with I would recommend picking up a grandstream. Again, they are cheap phones and prove it. After you understand the basics with this phone you would be best advised to never install grandstreams for production use and to learn how to install and provision Polycom or Snom phones. Polycom is regarded as the standard by which other SIP phones are measured. Cisco phones fall short of the average Polycom, unless you count good-looks. Expect to browse through over 2000-settings when provisioning these phones. BTW, Shoretel has polycom produce it's phones and rebrand them for their systems.
I do not recommend SIP-PSTN Gateways as they tend to add undue complexity to the system/dialplan and do not allow you to maintain good control over the call. Most of them are also sub-par as far as quality goes.
Feel free to ask me any other questions you may have regarding asterisk. In the spirit of OpenSource questions are encouraged and expected
