The first release candidate of Asterisk 1.2.0 is available

I use Asterisk in my voip / IP Telephony research, as well as quite a few other VOIP (open source) Solutions. I dealt with H323 for many a years now but I love SIP.
With all the support around asterisk VOIP PBX solution, I am debating my Choices. SER, OPENSER, SIPfoundry.
I use them all and my VOIP involvements never seems to end.
Anyway I am happy to promote any open source VOIP solutions. Asterisk is one of my favorites.
Kevin P. Fleming announced on the Asterisk-announce mailing list that Asterisk 1.2.0-rc1 Released!

The first release candidate of Asterisk 1.2.0 has been released! It is
available from the ftp.digium.com FTP servers, as well as the Digium CVS
servers (under the 'v1-2-0-rc1' tag).

This release includes a large number of improvements over beta2, including:

* Many bug fixes
* Documentation and sample configuration updates
* New 'stack' applications Gosub/Return/etc.


Asterisk
Asterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Who created Asterisk?
Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.