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How about xten, firefly, pulver, sjphone, sflphone, minisip, kphone, linphone, phonegaim? This is just the software I know of, there are probably many more that other people can add to the list. Which that would suit you depends of course on what your needs are.

Yes, theres windows, mac, and Linux clients.

For all I know (last visited their Web site just a few minutes ago), they currently sell only one product: the Virbiage 3010 ATA, a hardware solution.
I found out about their Firefly softphone from another source, and you can still download it from a 'hidden' (i.e. not reachable from their homepage) link.

SIP devices still have the problem of routing correctly through firewalls and the like. True P2P telephony is difficult with SIP, due to this. Skype gets around it by using their proprietary protocol. The much simpler and cleaner and far more open IAX2 protocol (a feature of the open source asterisk pbx) is being used by some devices to get around SIP limitations while still retaining (or exceeding) SIP voice quality. At least one IAX2 provider, firefly, gets it - (https://www.virbiage.com/products.php) calls to their network are automatically switched to the other user, getting the middleman out of the loop, and dramatically improving voice quality. Example - I place a call to a friend a block on vonage via vonage on comcast, and the packets get routed through about 17 routers, with a delay of 80ms - to get up the street. I place the same call via firefly - one router, and a delay of 25ms. Yes, theres windows, mac, and Linux clients. Also clients for most unixen, and several embedded devices....

If you were to run your own Asterisk server, you can always tell all your friends to download Firefly from http://www.virbiage.com/ which is a software phone not unlike the Skype client software, but instead of a closed proprietary protocol it supports SIP and IAX.

Your friends will then be able to call you directly on your Asterisk server and you will be able to call them on their softphone, all free of charge.

If they have their PC on a public IP address, SIP is OK, if they are behind a NAT (private IP address, eg 192.168.x.x or 10.x.x.x) then there is a chance they will not get incoming calls from you without fiddling. In that case IAX is your choice of protocol.

There main thing about Skype is that they have bolted on all the NAT traversal troubleshooting hacks of SIP and shrinkwrapped them bolted on to their protocol. This means that the end user doesn't have to worry about NAT as the software picks the most suitable workaround automatically. You may call that built-in NAT troubleshooting.

With standard SIP based solutions, you have to do the NAT traversal troubleshooting manually. Yet, with Asterisk you can always avoid those troubles altogether by using IAX instead of SIP. IAX was designed so it wouldn't need any workarounds for NAT in the first place. You may call that NAT troublefree.

With Asterisk you definitely get the better technology and more importantly freedom to choose equipment, protocols, codecs, service providers. With Skype you are totally locked in just like it used to be in Ma Bell's days.

Anyone with a trunk connection can announce whatever they want as the CID..

I definitely wouldn't put DISA (the ability to get an outside line) or anything like that on just CID authentication.

Firefly already exists.

Look at the IAX protocol that Asterisk PBX uses. Several devices are coming out now that talk this protocol.

Main selling points are:
#1 It works VERY well
#2 Only 1 port is ever used so NAT fowarding fixes all NAT issues
#3 Is a full PBX level intercommunication protocol so you can have any device using it do very advanced things that SIP and H323 only wish they could do well. (example... line in use indication for secretaries phones)

Virbiage is preparing to sell there FT201 based on IAX protocol and Digium (makers of Asterisk) are beginning mass production on their "IAXy" which is an ATA brick for analog phones.