One suggestion I have:<div><br></div><div>Would it be helpful to know the revision number of rc1 in the release notes? </div><div><br></div><div>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.</div>
<div><br></div><div>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. :)</div>
<div><br><div><div>Hope this helps!<br><div><br></div><div>James<br><br><div class="gmail_quote">On Mon, Oct 8, 2012 at 10:15 AM, Asterisk Development Team <span dir="ltr"><<a href="mailto:asteriskteam@digium.com" target="_blank">asteriskteam@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The Asterisk Development Team is pleased to announce the first release candidate<br>
of 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 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 DTLS-SRTP in chan_sip.<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-rc1" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>
--<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br>James Mortensen<br>Project Manager, VoiceCurve, Inc.<br>866-707-4590<br><a href="mailto:james.mortensen@voicecurve.com" target="_blank">james.mortensen@voicecurve.com</a><br>
<br>
</div></div></div></div>