<i>psc</i><br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Asterisk Development Team</b> <span dir="ltr"><<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>></span><br>
Date: 2012/8/10<br>Subject: [asterisk-dev] Asterisk 11.0.0-beta1 Now Available!<br>To: <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br><br><br>The Asterisk Development Team is pleased to announce the first beta release of<br>
Asterisk 11.0.0. This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases</a><br>
<br>
All interested users of Asterisk are encouraged to participate in the<br>
Asterisk 11 testing process. Please report any issues found to the issue<br>
tracker, <a href="https://issues.asterisk.org/jira" target="_blank">https://issues.asterisk.org/jira</a>. It is also very useful to see<br>
successful test reports. Please post those to the asterisk-dev mailing list.<br>
All Asterisk users are invited to participate in the #asterisk-testing channel<br>
on IRC to work together in testing the many parts of Asterisk.<br>
<br>
Asterisk 11 is the next major release series of Asterisk. It will be a Long<br>
Term Support (LTS) release, similar to Asterisk 1.8. For more information about<br>
support time lines for Asterisk releases, see the Asterisk versions page:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions</a><br>
<br>
For important information regarding upgrading to Asterisk 11, please see the<br>
Asterisk wiki:<br>
<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11</a><br>
<br>
A short list of new features includes:<br>
<br>
* A new channel driver named chan_motif has been added which provides support<br>
for Google Talk and Jingle in a single channel driver. This new channel<br>
driver includes support for both audio and video, RFC2833 DTMF, all codecs<br>
supported by Asterisk, hold, unhold, and ringing notification. It is also<br>
compliant with the current Jingle specification, current Google Jingle<br>
specification, and the original Google Talk protocol.<br>
<br>
* Support for the WebSocket transport for chan_sip.<br>
<br>
* SIP peers can now be configured to support negotiation of ICE candidates.<br>
<br>
* The app_page application now no longer depends on DAHDI or app_meetme. It<br>
has been re-architected to use app_confbridge internally.<br>
<br>
* Hangup handlers can be attached to channels using the CHANNEL() function.<br>
Hangup handlers will run when the channel is hung up similar to the h<br>
extension; however, unlike an h extension, a hangup handler is associated with<br>
the actual channel and will execute anytime that channel is hung up,<br>
regardless of where it is in the dialplan.<br>
<br>
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial<br>
allows you to execute a dialplan subroutine on a channel before a call is<br>
placed but after the application performing a dial action is invoked. This<br>
means that the handlers are executed after the creation of the caller/callee<br>
channels, but before any actions have been taken to actually dial the callee<br>
channels.<br>
<br>
* Log messages can now be easily associated with a certain call by looking at<br>
a new unique identifier, "Call Id". Call ids are attached to log messages for<br>
just about any case where it can be determined that the message is related<br>
to a particular call.<br>
<br>
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in<br>
Asterisk. Unlike traditional ACLs defined in specific module configuration<br>
files, Named ACLs can be shared across multiple modules.<br>
<br>
* The Hangup Cause family of functions and dialplan applications allow for<br>
inspection of the hangup cause codes for each channel involved in a call.<br>
This allows a dialplan writer to determine, for each channel, who hung up and<br>
for what reason(s).<br>
<br>
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()<br>
lets you set some of the configuration options from the general section<br>
of features.conf on a per-channel basis. FEATUREMAP() lets you customize<br>
the key sequence used to activate built-in features, such as blindxfer,<br>
and automon.<br>
<br>
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups<br>
and callgroups to be defined for several channel drivers.<br>
<br>
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.<br>
<br>
More information about the new features can be found on the Asterisk wiki:<br>
<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation</a><br>
<br>
A full list of all new features can also be found in the CHANGES file.<br>
<br>
<a href="http://svnview.digium.com/svn/asterisk/branches/11/CHANGES" target="_blank">http://svnview.digium.com/svn/asterisk/branches/11/CHANGES</a><br>
<br>
For a full list of changes in the current release, please see the ChangeLog.<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
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