Ótima notícia, pena que o Asterisk SCF deu uma pausa!<br><br><div class="gmail_quote">2012/9/20 Denis Galvão - Gmail <span dir="ltr"><<a href="mailto:denisgalvao@gmail.com" target="_blank">denisgalvao@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF"><div><br><br>Denis at mobile.</div><div><br>Begin forwarded message:<br><br></div><blockquote type="cite">
<div><b>From:</b> Asterisk Development Team <<a href="mailto:asteriskteam@digium.com" target="_blank">asteriskteam@digium.com</a>><br><b>Date:</b> 20 de setembro de 2012 16:40:00 BRT<br><b>To:</b> <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><br>
<b>Subject:</b> <b>[asterisk-dev] Asterisk 11.0.0-beta2 Now Available!</b><br><b>Reply-To:</b> Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a>><br>
<br></div></blockquote><div></div><blockquote type="cite"><div><span>The Asterisk Development Team is pleased to announce the second beta release of</span><br><span>Asterisk 11.0.0. This release is available for immediate download at</span><br>
<span><a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases</a></span><br><span></span><br><span>All interested users of Asterisk are encouraged to participate in the</span><br>
<span>Asterisk 11 testing process. Please report any issues found to the issue</span><br><span>tracker, <a href="https://issues.asterisk.org/jira" target="_blank">https://issues.asterisk.org/jira</a>. It is also very useful to see</span><br>
<span>successful test reports. Please post those to the asterisk-dev mailing list.</span><br><span>All Asterisk users are invited to participate in the #asterisk-testing channel</span><br><span>on IRC to work together in testing the many parts of Asterisk. </span><br>
<span></span><br><span>Asterisk 11 is the next major release series of Asterisk. It will be a Long</span><br><span>Term Support (LTS) release, similar to Asterisk 1.8. For more information about</span><br><span>support time lines for Asterisk releases, see the Asterisk versions page:</span><br>
<span><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions</a></span><br><span></span><br><span>For important information regarding upgrading to Asterisk 11, please see the</span><br>
<span>Asterisk wiki:</span><br><span></span><br><span><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></span><br>
<span></span><br><span>A short list of new features includes:</span><br><span></span><br><span>* A new channel driver named chan_motif has been added which provides support</span><br><span> for Google Talk and Jingle in a single channel driver. This new channel</span><br>
<span> driver includes support for both audio and video, RFC2833 DTMF, all codecs</span><br><span> supported by Asterisk, hold, unhold, and ringing notification. It is also</span><br><span> compliant with the current Jingle specification, current Google Jingle</span><br>
<span> specification, and the original Google Talk protocol.</span><br><span></span><br><span>* Support for the WebSocket transport for chan_sip.</span><br><span></span><br><span>* SIP peers can now be configured to support negotiation of ICE candidates.</span><br>
<span></span><br><span>* The app_page application now no longer depends on DAHDI or app_meetme. It</span><br><span> has been re-architected to use app_confbridge internally.</span><br><span></span><br><span>* Hangup handlers can be attached to channels using the CHANNEL() function.</span><br>
<span> Hangup handlers will run when the channel is hung up similar to the h</span><br><span> extension; however, unlike an h extension, a hangup handler is associated with</span><br><span> the actual channel and will execute anytime that channel is hung up,</span><br>
<span> regardless of where it is in the dialplan.</span><br><span></span><br><span>* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial</span><br><span> allows you to execute a dialplan subroutine on a channel before a call is</span><br>
<span> placed but after the application performing a dial action is invoked. This</span><br><span> means that the handlers are executed after the creation of the callee</span><br><span> channels, but before any actions have been taken to actually dial the callee</span><br>
<span> channels.</span><br><span></span><br><span>* Log messages can now be easily associated with a certain call by looking at</span><br><span> a new unique identifier, "Call Id". Call ids are attached to log messages for</span><br>
<span> just about any case where it can be determined that the message is related</span><br><span> to a particular call.</span><br><span></span><br><span>* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in</span><br>
<span> Asterisk. Unlike traditional ACLs defined in specific module configuration</span><br><span> files, Named ACLs can be shared across multiple modules.</span><br><span></span><br><span>* The Hangup Cause family of functions and dialplan applications allow for</span><br>
<span> inspection of the hangup cause codes for each channel involved in a call.</span><br><span> This allows a dialplan writer to determine, for each channel, who hung up and</span><br><span> for what reason(s).</span><br>
<span></span><br><span>* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()</span><br><span> lets you set some of the configuration options from the general section</span><br><span> of features.conf on a per-channel basis. FEATUREMAP() lets you customize</span><br>
<span> the key sequence used to activate built-in features, such as blindxfer,</span><br><span> and automon.</span><br><span></span><br><span>* Support for DTLS-SRTP in chan_sip.</span><br><span></span><br><span>* Support for named pickupgroups/callgroups, allowing any number of pickupgroups</span><br>
<span> and callgroups to be defined for several channel drivers.</span><br><span></span><br><span>* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.</span><br><span></span><br><span>More information about the new features can be found on the Asterisk wiki:</span><br>
<span></span><br><span><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></span><br><span></span><br><span>A full list of all new features can also be found in the CHANGES file.</span><br>
<span></span><br><span><a href="http://svnview.digium.com/svn/asterisk/branches/11/CHANGES" target="_blank">http://svnview.digium.com/svn/asterisk/branches/11/CHANGES</a></span><br><span></span><br><span>For a full list of changes in the current release, please see the ChangeLog.</span><br>
<span></span><br><span><a href="http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2</a></span><br>
<span></span><br><span>Thank you for your continued support of Asterisk!</span><br><span></span><br><span></span><br><span></span><br><span></span><br><span></span><br><span></span><br><span></span><br><span class="HOEnZb"><font color="#888888"><span></span><br>
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