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

[sean darcy]

On 10/08/2012 05:15 PM, Asterisk Development Team 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!
>
>

Thanks for all the great work.

We've started using the silk codec a lot for phone app voip. We've found 
it the most effective low bit rate (16K) codec. Could we get a release 
11 version of the silk codec in 
http://downloads.digium.com/pub/telephony/codec_silk/  ?

That way we could start messing with RC 1.

sean