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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Roadmap">Roadmap</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~mdavenport">Malcolm Davenport</a>
</h4>
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<h4>Changes (1)</h4>
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<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >* The [CHANGES|http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup] file in Asterisk lists new features that have been added to each version. <br> <br></td></tr>
<tr><td class="diff-changed-lines" >h1. Asterisk <span class="diff-changed-words">1<span class="diff-deleted-chars"style="color:#999;background-color:#fdd;text-decoration:line-through;">.1</span>0</span> <br></td></tr>
<tr><td class="diff-unchanged" > <br>*Committed Features* \- developed by Digium <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
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</div> <h4>Full Content</h4>
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<ul>
<li><a href='#Roadmap-RelatedLinks'>1. Related Links</a></li>
<li><a href='#Roadmap-Asterisk10'>2. Asterisk 10</a></li>
<li><a href='#Roadmap-FutureDevelopmentWishlist'>3. Future Development Wishlist</a></li>
<ul>
<li><a href='#Roadmap-%28P0%29'>3.1. (P0)</a></li>
<li><a href='#Roadmap-%28P1%29'>3.2. (P1)</a></li>
<li><a href='#Roadmap-%28P2%29'>3.3. (P2)</a></li>
<li><a href='#Roadmap-%28P3%29'>3.4. (P3)</a></li>
<li><a href='#Roadmap-%28P4%29'>3.5. (P4)</a></li>
<li><a href='#Roadmap-%28P5%29'>3.6. (P5)</a></li>
<li><a href='#Roadmap-%28P6%29'>3.7. (P6)</a></li>
<li><a href='#Roadmap-%28P7'>3.8. (P7</a></li>
<li><a href='#Roadmap-%28P8%29'>3.9. (P8)</a></li>
<li><a href='#Roadmap-%28P%3F%2CResearchRequired%29'>3.10. (P?, Research Required)</a></li>
</ul>
</ul></div>
<div class='panelMacro'><table class='warningMacro'><colgroup><col width='24'><col></colgroup><tr><td valign='top'><img src="/wiki/images/icons/emoticons/forbidden.gif" width="16" height="16" align="absmiddle" alt="" border="0"></td><td>The content of this page is a draft and is subject to change at any time.</td></tr></table></div>
<h1><a name="Roadmap-RelatedLinks"></a>1. Related Links</h1>
<ul>
<li>The <a href="https://issues.asterisk.org/roadmap_page.php" class="external-link" rel="nofollow">Roadmap Page</a> on the <a href="https://issues.asterisk.org" class="external-link" rel="nofollow">issue tracker</a> has information about various issues and patches that have been targeted for a specific Asterisk release.</li>
</ul>
<ul>
<li>The <a href="http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup" class="external-link" rel="nofollow">CHANGES</a> file in Asterisk lists new features that have been added to each version.</li>
</ul>
<h1><a name="Roadmap-Asterisk10"></a>2. Asterisk 10</h1>
<p><b>Committed Features</b> - developed by Digium</p>
<ul>
<li><a href="/wiki/display/AST/Media+Overhaul" title="Media Overhaul">Media Overhaul</a></li>
<li><a href="/wiki/display/AST/SIP+Security+Events" title="SIP Security Events">SIP Security Events</a></li>
<li><a href="/wiki/display/AST/T.38+Gateway" title="T.38 Gateway">T.38 Gateway</a></li>
<li><a href="/wiki/display/AST/Documentation+Improvements" title="Documentation Improvements">Documentation Improvements</a></li>
</ul>
<h1><a name="Roadmap-FutureDevelopmentWishlist"></a>3. Future Development Wishlist</h1>
<p>The items in this list primarily came from <a href="/wiki/display/AST/AstriDevCon+2010" title="AstriDevCon 2010">AstriDevCon 2010</a>.</p>
<h2><a name="Roadmap-%28P0%29"></a>3.1. (P0)</h2>
<p>P0 are committed projects that are going to get done.</p>
<ul>
<li><a href="/wiki/display/AST/T.38+Gateway" title="T.38 Gateway">T.38 Gateway</a> (Digium)</li>
<li>Performance of State Change Processing (Stefan Schmidt)
<ul>
<li>(work is already being done on this front)</li>
</ul>
</li>
<li>SIP path support (Olle)
<ul>
<li>(first generation of code exists, needs more work, simple patch, going to get it done, needs an extra field in astdb; helps when there are 2 or more load balancing proxies in front of asterisk, when you'd like the call to be able to get back to Asterisk; see <a href="https://reviewboard.asterisk.org/r/991/" class="external-link" rel="nofollow">https://reviewboard.asterisk.org/r/991/</a>)</li>
<li><a href="https://issues.asterisk.org/view.php?id=18223" class="external-link" rel="nofollow">https://issues.asterisk.org/view.php?id=18223</a></li>
</ul>
</li>
<li>Group variables (Kobaz)
<ul>
<li>(on review board, in progress)</li>
</ul>
</li>
<li>Pre-Dial (Kobaz)
<ul>
<li>(practically done, or something)</li>
</ul>
</li>
<li>Distributed extension state using SIP (Olle)
<ul>
<li>(resources in place, doing it, 1.4 done before Christmas, project pinana)</li>
</ul>
</li>
<li>Manager event docs (Paul Belanger)</li>
<li>Cross-platform documentation (Ben Klang)
<ul>
<li>(caveats for using Asterisk on operating system xyz; pull a PDF of the Wiki documentation into the source, don't forget to include basic installation information, and do it all in .txt - Ben)</li>
</ul>
</li>
<li>Fix libs to optionally init OpenSSL (Digium)
<ul>
<li>(or use existing tools; sort of a bug)</li>
</ul>
</li>
<li>Make ast_channel an opaque type (Digium)</li>
</ul>
<h2><a name="Roadmap-%28P1%29"></a>3.2. (P1)</h2>
<p>P1 is the highest priority.</p>
<ul>
<li><a href="/wiki/display/AST/Media+Overhaul" title="Media Overhaul">Codecs (SILK, Opal), Media Negotiation</a> (Digium)</li>
<li>RTCP (Olle)
<ul>
<li>Pinefrog; Work to be done - Ported to trunk, added to CEL</li>
</ul>
</li>
<li>Conferencing that supports a new magic media (Digium)
<ul>
<li>higher sampling rates</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P2%29"></a>3.3. (P2)</h2>
<ul class="alternate" type="square">
<li>Async DNS (TCP DNS and use a good resolver)</li>
<li>Named ACLs (deluxepine)</li>
<li><a href="/wiki/display/AST/SIP+Security+Events" title="SIP Security Events">SIP Security Events</a></li>
<li>Light weight means of holding NAT open in SIP (less complex than current qualify, Consider it done)</li>
<li>IPv6 for the restivus (IAX, Jabber/XMPP/Gtalk, Manager, etc.)</li>
<li>ConfBridge feature complete with MeetMe</li>
<li>Support sound file containers (matroska)</li>
</ul>
<h2><a name="Roadmap-%28P3%29"></a>3.4. (P3)</h2>
<ul>
<li>Who hung up? (there's a branch, shouldn't take too much time - Olle)</li>
<li>Unique identifier for filtering log data to a call
<ul>
<li>(finishing what was already begun w/ Clod's project, CLI filtering; should take a look at what Stephan from Unlimitel.ca's created)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P4%29"></a>3.5. (P4)</h2>
<ul>
<li>Multiple SIP Sockets
<ul>
<li>(Listen on multiple ports or on multiple interfaces, but not all; also set binding for RTP)...alternate idea / solution would be to make Asterisk capable of loading multiple SIP profiles, it might be easier</li>
</ul>
</li>
<li>Multiple DNS results
<ul>
<li>(need to be able to traverse a list of DNS results, rather than just getting back one result)</li>
</ul>
</li>
<li>ICE-lite
<ul>
<li>(no code, responding correctly to ICE connectivity checks (STUN multiplexed on the RTP port) and understanding the SDP); it makes NAT traversal work for clients that do ICE; also addressed lightweight NAT refresh)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P5%29"></a>3.6. (P5)</h2>
<ul>
<li>AstDB replacement
<ul>
<li>(realtime, there's code, nearly ready)</li>
</ul>
</li>
<li>SIP identity
<ul>
<li>(on reviewboard; needs to be forward ported; important for organizations w/ federated identities; a requirement for DTLS SRTP; not widely deployed)</li>
</ul>
</li>
<li>RTMP client channel driver</li>
</ul>
<h2><a name="Roadmap-%28P6%29"></a>3.7. (P6)</h2>
<ul>
<li>Structured identifiers for errors
<ul>
<li>(tag an error message with a unique string, specific to the error message and where it came from; should be alphanumeric to keep them short)</li>
</ul>
</li>
<li>AMI SetVar, Context limits
<ul>
<li>(there's code already...Olle has it)</li>
</ul>
</li>
<li>AMI filters on demand</li>
<li>DTLS SRTP
<ul>
<li>(not likely to be widely deployed in the next 12 months)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P7"></a>3.8. (P7</h2>
<ul>
<li>Asterisk register for XMPP account (Leif)</li>
<li>Write a Specification for AMI (not kobaz)</li>
<li>Multiple TLS server certs
<ul>
<li>(1 socket, requires support by OpenSSL; simpler to implement than multiple SIP profiles; don't know if any clients use it yet; needs more research)</li>
</ul>
</li>
</ul>
<h2><a name="Roadmap-%28P8%29"></a>3.9. (P8)</h2>
<ul class="alternate" type="square">
<li>Make resource modules that talk to DBs attempt reconnects</li>
<li>Apple's new file streaming format, derived from .m3u</li>
<li>Make MixMonitor and Monitor feature compatible</li>
</ul>
<h2><a name="Roadmap-%28P%3F%2CResearchRequired%29"></a>3.10. (P?, Research Required)</h2>
<ul>
<li>New app_queue (as if? no, seriously? talking about this scares Russell)</li>
<li>Identify and fix all bugs in AMI</li>
<li>Broadsoft or Dialog Info shared line appearance (SLA) support
<ul>
<li>(Tabled for later discussion)</li>
</ul>
</li>
<li>LDAP from within the dialplan
<ul>
<li>(we may already have it, needs research to see if the realtime driver does what's desired - Leif)</li>
</ul>
</li>
<li>Device state normalization</li>
<li>Anything DB over HTTP(s) with failover handling</li>
<li>Use a channel as a MoH Source</li>
<li>Kill Masquerades</li>
<li>Bridging thread pool</li>
<li>Threadify chan_sip</li>
<li>Export ISDN ROSE information up to Asterisk channels</li>
</ul>
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