[asterisk-users] Asterisk 11.0.0-rc1 Now Available!

Mon Oct 8 19:23:10 CDT 2012

One suggestion I have:

Would it be helpful to know the revision number of rc1 in the release
notes?

I'm on a patched version of Asterisk from Doubango to deal with Chrome's
non-standard ICE candidates, and unless this is included in rc1 (meaning
rc1 is newer than what I have and also deals with Chrome's ICE issues) then
I probably wouldn't upgrade.  Also, I would prefer to check out from source
but don't know the revision number to use.

If I'm the only one that would benefit from this, then no worries, I'll
deal with it. But if others would benefit from seeing a revision
number/checking out from SVN, then maybe consider adding this to the
release notes. :)

Hope this helps!

James

On Mon, Oct 8, 2012 at 10:15 AM, Asterisk Development Team <
asteriskteam at digium.com> wrote:

> The Asterisk Development Team is pleased to announce the first release
> candidate
> of Asterisk 11.0.0.  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 11 testing process.  Please report any issues found to the issue
> tracker, https://issues.asterisk.org/jira.  It is also very useful to see
> successful test reports.  Please post those to the asterisk-dev mailing
> list.
> All Asterisk users are invited to participate in the #asterisk-testing
> channel
> on IRC to work together in testing the many parts of Asterisk.
>
> Asterisk 11 is the next major release series of Asterisk.  It will be a
> Long
> Term Support (LTS) release, similar to Asterisk 1.8.  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 11, please see
> the
> Asterisk wiki:
>
> https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
>
> A short list of new features includes:
>
> * A new channel driver named chan_motif has been added which provides
> support
>   for Google Talk and Jingle in a single channel driver.  This new channel
>   driver includes support for both audio and video, RFC2833 DTMF, all
> codecs
>   supported by Asterisk, hold, unhold, and ringing notification. It is also
>   compliant with the current Jingle specification, current Google Jingle
>   specification, and the original Google Talk protocol.
>
> * Support for the WebSocket transport for chan_sip.
>
> * SIP peers can now be configured to support negotiation of ICE candidates.
>
> * The app_page application now no longer depends on DAHDI or app_meetme. It
>   has been re-architected to use app_confbridge internally.
>
> * Hangup handlers can be attached to channels using the CHANNEL() function.
>   Hangup handlers will run when the channel is hung up similar to the h
>   extension; however, unlike an h extension, a hangup handler is
> associated with
>   the actual channel and will execute anytime that channel is hung up,
>   regardless of where it is in the dialplan.
>
> * Added pre-dial handlers for the Dial and Follow-Me applications.
>  Pre-dial
>   allows you to execute a dialplan subroutine on a channel before a call is
>   placed but after the application performing a dial action is invoked.
> This
>   means that the handlers are executed after the creation of the callee
>   channels, but before any actions have been taken to actually dial the
> callee
>   channels.
>
> * Log messages can now be easily associated with a certain call by looking
> at
>   a new unique identifier, "Call Id".  Call ids are attached to log
> messages for
>   just about any case where it can be determined that the message is
> related
>   to a particular call.
>
> * Introduced Named ACLs as a new way to define Access Control Lists (ACLs)
> in
>   Asterisk. Unlike traditional ACLs defined in specific module
> configuration
>   files, Named ACLs can be shared across multiple modules.
>
> * The Hangup Cause family of functions and dialplan applications allow for
>   inspection of the hangup cause codes for each channel involved in a call.
>   This allows a dialplan writer to determine, for each channel, who hung
> up and
>   for what reason(s).
>
> * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
>   lets you set some of the configuration options from the general section
>   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
>   the key sequence used to activate built-in features, such as blindxfer,
>   and automon.
>
> * Support for DTLS-SRTP in chan_sip.
>
> * Support for named pickupgroups/callgroups, allowing any number of
> pickupgroups
>   and callgroups to be defined for several channel drivers.
>
> * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event
> Framework.
>
> More information about the new features can be found on the Asterisk wiki:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
>
> A full list of all new features can also be found in the CHANGES file.
>
> http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
>
> For a full list of changes in the current release, please see the
> ChangeLog.
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1
>
> Thank you for your continued support of Asterisk!
>
>
> --
> _____________________________________________________________________
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-- 
James Mortensen
Project Manager, VoiceCurve, Inc.
866-707-4590
james.mortensen at voicecurve.com
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