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    <h2><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Developer+Call+-+07142011+-+1700+EST">Asterisk Developer Call - 07142011 - 1700 EST</a></h2>
    <h4>Page  <b>added</b> by             <a href="https://wiki.asterisk.org/wiki/display/~bmj">Bryan M. Johns</a>
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         <p><b>Digium Developer Conference Bridge:</b></p>

<p>Bridge:&nbsp; 15071</p>

<p>PIN:&nbsp; 35806</p>

<p><b>Proposed Agenda:</b></p>

<p>Welcome &amp; Announcements:&nbsp; Bryan Johns / Russell Bryant</p>


<ul>
        <li>Introduction of Digium participants</li>
</ul>


<p>Old Business:</p>

<p>None</p>

<p>New Business:</p>
<ul>
        <li>AstriCon 2011</li>
        <li>Asterisk 1.8 development update</li>
        <li>Asterisk 1.10 development update</li>
</ul>


<p>Asterisk 1.10 Beta Release Announcement for Reference</p>

<p>--<del>BEGIN PASTE</del>--<br/>
The Asterisk Development Team is pleased to announce the release of Asterisk 1.10.0-beta1. This release is available for immediate download at <a href="http://downloads.asterisk.org/pub/telephony/asterisk/" class="external-link" rel="nofollow">http://downloads.asterisk.org/pub/telephony/asterisk/</a></p>

<p>All interested users of Asterisk are encouraged to participate in the 1.10 testing process. Please report any issues found to the issue tracker, <a href="https://issues.asterisk.org/jira" class="external-link" rel="nofollow">https://issues.asterisk.org/jira</a>. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.</p>

<p>All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases.</p>

<p>Asterisk 1.10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page.</p>

<p><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions" class="external-link" rel="nofollow">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions</a></p>

<p>A short list of included features includes:</p>

<ul>
        <li>T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).</li>
        <li>Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.</li>
        <li>New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8KHz-196KHz</li>
        <li>Addition of video_mode option in confbridge.conf for the addition of basic video conferencing in the ConfBridge() dialplan application.</li>
        <li>Support for defining hints has been added to pbx_lua.</li>
</ul>


<ul>
        <li>Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).</li>
        <li>Much, much more!</li>
</ul>


<p>A full list of new features can be found in the CHANGES file.</p>

<p><a href="http://svn.digium.com/view/asterisk/branches/1.10/CHANGES?view=checkout" class="external-link" rel="nofollow">http://svn.digium.com/view/asterisk/branches/1.10/CHANGES?view=checkout</a></p>

<p>For a full list of changes in the current release, please see the ChangeLog:</p>

<p><a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.10.0-beta1" class="external-link" rel="nofollow">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.10.0-beta1</a></p>

<p>Thank you for your continued support of Asterisk!<br/>
--<del>END PASTE</del>--</p>

<p>Open Floor</p>

<p>Notes:</p>
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