[asterisk-users] Asterisk 12.0.0-beta2 Now Available!

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 12.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 12 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. All Asterisk users are invited to
participate in the #asterisk-bugs channel to help communicate issues found to
the Asterisk developers. It is also very useful to see successful test reports.
Please post those to the asterisk-dev mailing list (http://lists.digium.com).

Asterisk 12 is the next major release series of Asterisk. It will be a Standard
release, similar to Asterisk 10.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 12, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12

As Asterisk 12 has a more flexible feature policy, a number of new features
have been included in the second beta release that were not present in
the first beta release. These include:

* Numerous new features and improvements to ARI, including the ability to
  manipulate and query device state, snoop on channels (similar to the ChanSpy
  dialplan application), and improvements to the overall consistency of the
  RESTful API.

* New AMI commands for the PJSIP stack, including:
  - PJSIPShowEndpoint/PJSIPShowEndpoints for querying endpoint information
  - PJSIPShowRegistrationsInbound/PJSIPShowRegistrationsOutbound for querying
    registration information
  - PJSIPShowSubscriptionsInbound/PJSIPShowSubscriptionsOutbound for querying
    subscription information.

* Major improvements to sip_to_pjsip.py, a sip.conf to pjsip.conf conversion
  script. Users migrating to the PJSIP stack can use this script to aid in the
  migration process from an existing sip.conf configuration file.

* Standardization of the PJSIP configuration parameters. The configuration
  option schema for pjsip.conf now uses snake case for all of its options. This
  will impact current beta testers, as options such as 'fromuser' will now be
  'from_user', etc. Please verify your options against the documentation on the
  Asterisk wiki at: https://wiki.asterisk.org/wiki/x/ZwWUAQ

* Many bug fixes reported by the Asterisk community. Thank you for your
  collaboration and participation in testing the first Asterisk 12 beta release!

For more information on the Asterisk 12 release policy, please see the Asterisk
wiki:

https://wiki.asterisk.org/wiki/x/1YRHAQ

In addition to these improvements in the second beta release, Asterisk 12
contains many new major freatures. A short list of some of these features
includes:

* A new SIP channel driver and accompanying SIP stack named chan_pjsip has been
  added. This new channel driver is based on the PJSIP SIP stack by Teluu. It
  includes support for the vast majority of features currently in chan_sip,
  as well as numerous architectural improvements that alleviate pain points
  present in the legacy SIP channel driver. Users who wish to use the new SIP
  channel driver are encouraged to read the instructions on installing and
  configuring PJSIP for Asterisk on the Asterisk wiki at
  https://wiki.asterisk.org/wiki/x/J4GLAQ. Detailed instructions on configuring
  the new SIP stack in Asterisk can be found on the Asterisk wiki as well, at
  https://wiki.asterisk.org/wiki/x/hYCLAQ. Test reports of successful use of
  chan_pjsip, with endpoint details, in addition to bug reports, are most
  welcome.

* The Asterisk REST Interface (ARI) has been added. This interface lets
  external systems harness the telephony primitives within Asterisk to develop
  their own communications applications. Communication with Asterisk is done
  through a RESTful interface, while asynchronous events from Asterisk are
  encoded in JSON and sent via a WebSocket. More information on ARI can be found
  at https://wiki.asterisk.org/wiki/x/lYBbAQ

* Major standardization of the Asterisk Manager Interface and its events have
  occurred within this version. In particular, the names of Asterisk channels
  no longer change and are stable throughout the lifetime of the channel.
  More information on the changes in AMI can be seen in the AMI 1.4
  Specification at https://wiki.asterisk.org/wiki/x/dAFRAQ

* All bridging within Asterisk is now performed using the Asterisk Bridging API,
  which previously was only used by the ConfBridge application. This affords
  Asterisk users greater stability, and has resulted in the abstraction of
  channel masquerades, renaming, and other internal implementation details. It
  also allows for the seamless transition between two-party and multi-party
  bridges using core features.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/12/CHANGES

For a full list of changes in the current release, please see the ChangeLog.

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-beta2

Thank you for your continued support of Asterisk!