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
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