Of course, if you want to enjoy the highest quality audio then use FreeSWITCH because it supports audio codecs at multiple sampling rates and bit rates. It also has a Skype module (recently renamed "Skypopen" from "Skypiax") that will piggy-back off of the Skype client and handle the high def audio from Skype's SILK codec.
Other things you can do with Asterisk (or FreeSWITCH) would be things like answering incoming calls and handling them differently based upon the caller ID presented. Known telemarketers could be sent to so-called "torture" scripts that play fun little pranks on those calling. You could also route certain callers to a follow-me setup where, so if your kids are calling you could have it ring to your cell phone after ringing the home phone for 12 seconds, but if your boss is calling you could route it to a message saying, "Your call is very important to me, please leave a message and I will return your call at my earliest convenience..."
The are many things you can do with OSS telephony. I'm a HUGE fan of FreeSWITCH but that doesn't that I think you shouldn't look at Asterisk, YATE, OpenSIPS, etc. if you are curious about telephony.
Enjoy,
MC