The Asterisk Development Team is pleased to announce the first alpha 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).
The first preliminary test release of Asterisk 12 is an alpha release, not a beta release. Due to the size and scope of the changes in Asterisk 12, both an alpha test cycle and a beta test cycle will be performed. While users are encouraged to participate in both test cycles, users who choose to participate in the alpha release testing should understand that an alpha release has not undergone all of the community testing that a beta release goes through.
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:
A short list of some of the new major 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. Detailed instructions on configuring the new SIP stack in Asterisk can be found on the Asterisk wiki as wel. Test reports of successful use of chan_pjsip, with endpoint details, in addition to bug reports, are most welcome.
- The Asterisk RESTful 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 REST 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.
- 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:
A full list of all new features can also be found in the CHANGES file.
For a full list of changes in the current release, please see the ChangeLog.
Thank you for your continued support of Asterisk!