<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><br><div>Begin forwarded message:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Asterisk Development Team <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>[asterisk-dev] Asterisk 11.0.0-beta1 Now Available!</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">10 de agosto de 2012 16:25:00 BRT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br></span></div><br><div>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">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, https://issues.asterisk.org/jira. 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>https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions<br><br>For important information regarding upgrading to Asterisk 11, please see the<br>Asterisk wiki:<br><br>https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11<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>https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation<br><br>A full list of all new features can also be found in the CHANGES file.<br><br>http://svnview.digium.com/svn/asterisk/branches/11/CHANGES<br><br>For a full list of changes in the current release, please see the ChangeLog.<br><br>http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1<br><br>Thank you for your continued support of Asterisk!<br><br><br><br><br><br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-dev<br></div></blockquote></div><br></body></html>