The second beta 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-beta2' tag). This beta includes a large number of improvements over beta1, including: * Many, many bug fixes * Documentation and sample configuration updates * Vastly improved presence/subscription support in the SIP channel driver * A new (experimental) mISDN channel driver * A new monitoring application (MixMonitor) * More portability fixes for non-Linux platforms * New dialplan functions replacing old applications * Significant deadlock and performance upgrades for the Manager interface * An upgrade to the 'new' dialplan expression parser for all users * New Zaptel echo cancellers with improved performance * Support for the latest OSP toolkit from TransNexus * Support user-controlled volume adjustment in MeetMe application * More dialplan applications now return status variables instead of priority jumping * Much more powerful ENUM support in the dialplan * SIP domain support for authentication and virtual hosting * Many PRI protocol updates and fixes, including more complete Q.SIG support * New applications: Pickup() and Page() * ... and many more I'm sure I've missed! We ask all interested community members to download and install the beta release (on a non-production server) and report their findings via our bug tracker. Please be sure to read the UPGRADE.txt file in the distribution before upgrading your server, as there are a large number of changes that you will need to be aware of (some of them are not backwards compatible with the 1.0.x releases). We want to extend our thanks to all the community members whose contributions have made this release possible; without their support, testing and other involvement we would not have reached this milestone so soon!
We're having a meeting at 10am GMT in the English conference rooms (691) to discuss the new beta (installation problems/requests etc). The conference rooms (IAX2) are available at http://freevoip.gedameurope.com If anyone would like a free hosted conference room for an Asterisk related meeting, please don't hesitate to let us know. If you don't want to join up with FreeVoip you can just dial (IAX2/freevoip.gedameurope.com/691) We've got dev's standing by to help people! :D Let's get this beta process done and onto 1.2!! -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
I just went thru the docs and UPGRADE file. Is there any other place that has a more detailed description of the changes/additions on this new version? For example, new apps, changes in the dialplan, "New dialplan functions replacing old applications"? |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Kevin P. Fleming |Sent: Tuesday, November 01, 2005 12:43 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Asterisk 1.2.0-beta2 Released | |The second beta 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-beta2' tag). | |This beta includes a large number of improvements over beta1, |including: | | * Many, many bug fixes | * Documentation and sample configuration updates | * Vastly improved presence/subscription support in the SIP |channel driver | * A new (experimental) mISDN channel driver | * A new monitoring application (MixMonitor) | * More portability fixes for non-Linux platforms | * New dialplan functions replacing old applications | * Significant deadlock and performance upgrades for the |Manager interface | * An upgrade to the 'new' dialplan expression parser for all users | * New Zaptel echo cancellers with improved performance | * Support for the latest OSP toolkit from TransNexus | * Support user-controlled volume adjustment in MeetMe application | * More dialplan applications now return status variables |instead of priority jumping | * Much more powerful ENUM support in the dialplan | * SIP domain support for authentication and virtual hosting | * Many PRI protocol updates and fixes, including more |complete Q.SIG support | * New applications: Pickup() and Page() | * ... and many more I'm sure I've missed! | |We ask all interested community members to download and |install the beta release (on a non-production server) and |report their findings via our bug tracker. Please be sure to |read the UPGRADE.txt file in the distribution before upgrading |your server, as there are a large number of changes that you |will need to be aware of (some of them are not backwards |compatible with the 1.0.x releases). | |We want to extend our thanks to all the community members |whose contributions have made this release possible; without |their support, testing and other involvement we would not have |reached this milestone so soon! | | |_______________________________________________ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
On Tuesday 01 November 2005 08:43, Kevin P. Fleming wrote:> The second beta 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-beta2' tag). >I think you forgot to place it to ftp server. http://ftp.digium.com/pub/asterisk/asterisk-1.2.0-beta2.tar.gz gives NOT FOUND Regards, Roman
Roman wrote:> I think you forgot to place it to ftp server. > http://ftp.digium.com/pub/asterisk/asterisk-1.2.0-beta2.tar.gz > gives NOT FOUNDThis is being fixed right now... one of the FTP servers was out of sync.