[svn-commits] bebuild: tag 11.1.0 r377508 - /tags/11.1.0/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Dec 9 19:33:03 CST 2012
Author: bebuild
Date: Sun Dec 9 19:33:00 2012
New Revision: 377508
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377508
Log:
Importing release summary for 11.1.0 release.
Added:
tags/11.1.0/asterisk-11.1.0-summary.html (with props)
tags/11.1.0/asterisk-11.1.0-summary.txt (with props)
Added: tags/11.1.0/asterisk-11.1.0-summary.html
URL: http://svnview.digium.com/svn/asterisk/tags/11.1.0/asterisk-11.1.0-summary.html?view=auto&rev=377508
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--- tags/11.1.0/asterisk-11.1.0-summary.html (added)
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+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-11.1.0</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-11.1.0</h3>
+<h3 align="center">Date: 2012-12-09</h3>
+<h3 align="center"><asteriskteam at digium.com></h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+ <li><a href="#summary">Summary</a></li>
+ <li><a href="#contributors">Contributors</a></li>
+ <li><a href="#issues">Closed Issues</a></li>
+ <li><a href="#commits">Other Changes</a></li>
+ <li><a href="#diffstat">Diffstat</a></li>
+</ol>
+<hr/>
+<a name="summary"><h2 align="center">Summary</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p>
+<p>The data in this summary reflects changes that have been made since the previous release, asterisk-11.0.0.</p>
+<hr/>
+<a name="contributors"><h2 align="center">Contributors</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
+<table width="100%" border="0">
+<tr>
+<td width="33%"><h3>Coders</h3></td>
+<td width="33%"><h3>Testers</h3></td>
+<td width="33%"><h3>Reporters</h3></td>
+</tr>
+<tr valign="top">
+<td>
+6 bebuild<br/>
+6 jrose<br/>
+6 rmudgett<br/>
+5 file<br/>
+5 mjordan<br/>
+4 jcolp<br/>
+3 Corey Farrell<br/>
+3 kmoore<br/>
+3 mmichelson<br/>
+2 JoshE<br/>
+2 Richard Miller<br/>
+2 sruffell<br/>
+2 wdoekes<br/>
+1 Bryan Walters<br/>
+1 Daniel O'Connor<br/>
+1 David Chappell<br/>
+1 elguero<br/>
+1 feyfre<br/>
+1 Guenther Kelleter<br/>
+1 igorg<br/>
+1 jbigelow<br/>
+1 lathama<br/>
+1 twilson<br/>
+1 tzafrir<br/>
+1 wedhorn<br/>
+</td>
+<td>
+1 Dmitry Burilov<br/>
+1 mjordan<br/>
+1 rmudgett<br/>
+1 Thomas Arimont<br/>
+</td>
+<td>
+3 coreyfarrell<br/>
+3 kmoore<br/>
+3 mjordan<br/>
+2 tomaso<br/>
+2 ulogic<br/>
+1 chappell<br/>
+1 danjenkins<br/>
+1 daren<br/>
+1 darius<br/>
+1 deniz<br/>
+1 feyfre<br/>
+1 gamegamer43<br/>
+1 gkelleter<br/>
+1 jbigelow<br/>
+1 licedey<br/>
+1 n8ideas<br/>
+1 netaskd<br/>
+1 pciccone<br/>
+1 sruffell<br/>
+1 stocksy<br/>
+1 tblancher<br/>
+1 vilius365<br/>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="issues"><h2 align="center">Closed Issues</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
+<h3>Category: Applications/app_confbridge</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19448">ASTERISK-19448</a>: ConfBridge crashes Asterisk when no timing module loaded.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375506">375506</a><br/>
+Reporter: feyfre<br/>
+Coders: feyfre<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20601">ASTERISK-20601</a>: Confbridge recording does not work<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375471">375471</a><br/>
+Reporter: vilius365<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Applications/app_queue</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20623">ASTERISK-20623</a>: App_queue doesn't increment number of busy agent in certain situations<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375469">375469</a><br/>
+Reporter: gamegamer43<br/>
+Coders: Bryan Walters<br/>
+<br/>
+<h3>Category: Channels/chan_motif</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20554">ASTERISK-20554</a>: Outgoing calls fail to establish audio due to ICE negotiation failures<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374850">374850</a><br/>
+Reporter: mjordan<br/>
+Coders: jcolp<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18203">ASTERISK-18203</a>: Problems with NAT on realtime peers (and maybe static ones)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375437">375437</a><br/>
+Reporter: daren<br/>
+Coders: JoshE<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20487">ASTERISK-20487</a>: Failure to have OpenSSL w/ SRTP support results in confusing error message<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374755">374755</a><br/>
+Reporter: mjordan<br/>
+Coders: jcolp<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20487">ASTERISK-20487</a>: Failure to have OpenSSL w/ SRTP support results in confusing error message<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374756">374756</a><br/>
+Reporter: mjordan<br/>
+Coders: jcolp<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20561">ASTERISK-20561</a>: Asterisk 1.8 allows the # character in SIP URI, 10 and higher versions do not - need to document in UPGRADE.txt possibly other places?<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375847">375847</a><br/>
+Reporter: deniz<br/>
+Coders: jrose<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20572">ASTERISK-20572</a>: Realtime Peers behind NAT are Set to RFC1918 private address after sip reload<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375437">375437</a><br/>
+Reporter: n8ideas<br/>
+Coders: JoshE<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Messaging</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20545">ASTERISK-20545</a>: chan_sip loads too early because of exposed global symbols<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374842">374842</a><br/>
+Reporter: kmoore<br/>
+Coders: mmichelson<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20611">ASTERISK-20611</a>: sip registery lost after sip reload<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375575">375575</a><br/>
+Reporter: licedey<br/>
+Coders: jrose<br/>
+<br/>
+<h3>Category: Channels/chan_sip/TCP-TLS</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20212">ASTERISK-20212</a>: Deadlock / TCP SIP Stack<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374914">374914</a><br/>
+Reporter: pciccone<br/>
+Coders: mmichelson<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20559">ASTERISK-20559</a>: SIP TCP/TLS: When checking the CA certificate fails, the call still goes through<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375148">375148</a><br/>
+Reporter: kmoore<br/>
+Coders: kmoore<br/>
+<br/>
+<h3>Category: Codecs/codec_gsm</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20457">ASTERISK-20457</a>: GSM encoding is not thread safe<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375288">375288</a><br/>
+Reporter: ulogic<br/>
+Coders: Richard Miller<br/>
+<br/>
+<h3>Category: Contrib/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20260">ASTERISK-20260</a>: Increase robustness of ast_tls_cert<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375327">375327</a><br/>
+Reporter: darius<br/>
+Coders: Daniel O'Connor<br/>
+<br/>
+<h3>Category: Core/AstDB</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20647">ASTERISK-20647</a>: [patch] Failure to cleanup SQLite3 statements during exit causes call to sqlite3_close to fail; leaks memory<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375763">375763</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: Corey Farrell<br/>
+<br/>
+<h3>Category: Core/BuildSystem</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20483">ASTERISK-20483</a>: Allow Asterisk to report git SHAs in version string.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375191">375191</a><br/>
+Reporter: sruffell<br/>
+Coders: sruffell<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20648">ASTERISK-20648</a>: [patch] - Memory leaks in xmldoc<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375756">375756</a><br/>
+Reporter: coreyfarrell<br/>
+Testers: mjordan<br/>
+Coders: Corey Farrell<br/>
+<br/>
+<h3>Category: Core/ManagerInterface</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20544">ASTERISK-20544</a>: action_originate called via ast_hook_send_action causes a segfault<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374792">374792</a><br/>
+Reporter: kmoore<br/>
+Coders: kmoore<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20646">ASTERISK-20646</a>: [patch] - manager_shutdown fails to completely shutdown AMI and leaks memory<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375795">375795</a><br/>
+Reporter: coreyfarrell<br/>
+Coders: Corey Farrell<br/>
+<br/>
+<h3>Category: Core/PBX</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20455">ASTERISK-20455</a>: dialplan fails to run the invalid "i" extension due to an uninitialized variable dat_exten in main/pbx.c<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374771">374771</a><br/>
+Reporter: ulogic<br/>
+Coders: Richard Miller<br/>
+<br/>
+<h3>Category: Documentation</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-14435">ASTERISK-14435</a>: [patch] Add option and description to chan_dahdi.conf.sample<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374729">374729</a><br/>
+Reporter: jbigelow<br/>
+Coders: jbigelow, sruffell<br/>
+<br/>
+<h3>Category: Features/Parking</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19382">ASTERISK-19382</a>: Park() ignores 'r' option, plays default MOH instead.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375390">375390</a><br/>
+Reporter: stocksy<br/>
+Coders: rmudgett<br/>
+<br/>
+<h3>Category: PBX/pbx_realtime</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18203">ASTERISK-18203</a>: Problems with NAT on realtime peers (and maybe static ones)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375437">375437</a><br/>
+Reporter: daren<br/>
+Coders: JoshE<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20572">ASTERISK-20572</a>: Realtime Peers behind NAT are Set to RFC1918 private address after sip reload<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375437">375437</a><br/>
+Reporter: n8ideas<br/>
+Coders: JoshE<br/>
+<br/>
+<h3>Category: PBX/pbx_spool</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17231">ASTERISK-17231</a>: [patch] unopenable spool files not deleted<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374708">374708</a><br/>
+Reporter: chappell<br/>
+Coders: David Chappell<br/>
+<br/>
+<h3>Category: Resources/res_calendar_ews</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19738">ASTERISK-19738</a>: Calendar EWS does not attempt to extract the Body element in a CalendarItem and populate the description event field<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375532">375532</a><br/>
+Reporter: netaskd<br/>
+Testers: Dmitry Burilov<br/>
+Coders: twilson<br/>
+<br/>
+<h3>Category: Resources/res_http_websocket</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20631">ASTERISK-20631</a>: Unable to connect via WebRTC<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375559">375559</a><br/>
+Reporter: danjenkins<br/>
+Coders: jcolp<br/>
+<br/>
+<hr/>
+<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
+<table width="100%" border="1">
+<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374833">374833</a></td><td>file</td><td>Consider the Google Talk content stanza name (jin:content) valid.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374851">374851</a></td><td>file</td><td>Remove code that should not have gotten in.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20554">ASTERISK-20554</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374877">374877</a></td><td>file</td><td>Fix a bug where audio on Google Voice would not work due to ignoring candidates.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374932">374932</a></td><td>kmoore</td><td>Avoid a segfault on invalid format names</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=374995">374995</a></td><td>tzafrir</td><td>Update config.guess and config.sub: 2012-10-10</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375016">375016</a></td><td>igorg</td><td></td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375027">375027</a></td><td>mmichelson</td><td>Fix some potential misuses of ast_str in the code.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375051">375051</a></td><td>file</td><td>Remove a log message that was left in accidentally from call-id logging development.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375079">375079</a></td><td>wdoekes</td><td>Update sip_request_call SIP dial string documentation.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375113">375113</a></td><td>wdoekes</td><td>Fixes to the fd-oriented SIP TCP reads.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375219">375219</a></td><td>jrose</td><td>app_queue: Make ordering of rrmemory/rrordered persist over add/remove members</td>
+<td><a href="https://issues.asterisk.org/jira/browse/AST-989">AST-989</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375247">375247</a></td><td>jrose</td><td>app_queue: add upgrade notes for 375216</td>
+<td><a href="https://issues.asterisk.org/jira/browse/AST-989">AST-989</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375486">375486</a></td><td>jrose</td><td>mixmonitor: Add a test event</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375613">375613</a></td><td>elguero</td><td>Fix Wrong Result In Debug Message For SDP Origin Processing</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375627">375627</a></td><td>rmudgett</td><td>Multiple revisions 375519-375524</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375660">375660</a></td><td>wedhorn</td><td>Fix for chan_skinny leaving RTP ports open</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375661">375661</a></td><td>rmudgett</td><td>Things don't need to be that const.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375702">375702</a></td><td>lathama</td><td>Doxygen Updates</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20259">ASTERISK-20259</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375729">375729</a></td><td>mjordan</td><td>Prevent multiple CDR batches from conflicting when scheduling the CDR write</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375798">375798</a></td><td>mjordan</td><td>Only deref a reserved gateway session if we actually reserved one</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375802">375802</a></td><td>mjordan</td><td>Don't attempt to purge sessions when no sessions exist</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375864">375864</a></td><td>rmudgett</td><td>Add safety NULL pointer check in module user references.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375895">375895</a></td><td>mjordan</td><td>Refactor ast_timer_ack to return an error and handle the error in timer users</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20032">ASTERISK-20032</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=375925">375925</a></td><td>file</td><td>Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377293">377293</a></td><td>bebuild</td><td>Create 11.1.0-rc2</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377299">377299</a></td><td>bebuild</td><td>Merge r377259 for 11.1.0-rc2</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377302">377302</a></td><td>bebuild</td><td>Importing release summary for 11.1.0-rc2 release.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377318">377318</a></td><td>bebuild</td><td>Create 11.1.0-rc3</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377323">377323</a></td><td>bebuild</td><td>Merge r376870 for 11.1.0-rc3</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/11?view=revision&revision=377328">377328</a></td><td>bebuild</td><td>Importing release summary for 11.1.0-rc3 release.</td>
+<td></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+ChangeLog | 11
+Makefile | 2
+README | 2
+UPGRADE.txt | 16
+apps/app_confbridge.c | 4
+apps/app_dial.c | 12
+apps/app_mixmonitor.c | 8
+apps/app_queue.c | 159 +++++
+apps/app_voicemail.c | 5
+asterisk-11.1.0-rc2-summary.html | 60 --
+asterisk-11.1.0-rc2-summary.txt | 88 ---
+asterisk-11.1.0-rc3-summary.html | 62 ++
+asterisk-11.1.0-rc3-summary.txt | 94 +++
+bridges/bridge_softmix.c | 12
+build_tools/make_version | 106 +++
+channels/chan_dahdi.c | 17
+channels/chan_iax2.c | 36 -
+channels/chan_local.c | 21
+channels/chan_misdn.c | 2
+channels/chan_motif.c | 30 -
+channels/chan_sip.c | 1041 +++++++++++++++++++++++++++++++++-----
+channels/chan_sip.exports.in | 6
+channels/chan_skinny.c | 39 -
+channels/chan_unistim.c | 20
+channels/misdn/isdn_lib.c | 218 +++----
+channels/misdn/isdn_lib.h | 3
+codecs/gsm/src/code.c | 3
+config.guess | 279 +++++-----
+config.sub | 236 ++++++--
+configs/chan_dahdi.conf.sample | 27
+configs/sip.conf.sample | 16
+configure.ac | 1
+contrib/scripts/ast_tls_cert | 41 +
+funcs/func_jitterbuffer.c | 5
+include/asterisk/autoconfig.h.in | 13
+include/asterisk/doxyref.h | 71 +-
+include/asterisk/sip_api.h | 27
+include/asterisk/strings.h | 22
+include/asterisk/tcptls.h | 6
+include/asterisk/timing.h | 9
+main/app.c | 1
+main/ccss.c | 20
+main/cdr.c | 12
+main/channel.c | 16
+main/db.c | 36 +
+main/features.c | 13
+main/format_pref.c | 4
+main/loader.c | 12
+main/manager.c | 60 +-
+main/pbx.c | 3
+main/sip_api.c | 60 ++
+main/tcptls.c | 30 -
+main/timing.c | 16
+main/xmldoc.c | 7
+makeopts.in | 1
+pbx/pbx_spool.c | 301 +++++-----
+res/res_calendar_ews.c | 18
+res/res_fax.c | 4
+res/res_fax_spandsp.c | 7
+res/res_http_websocket.exports.in | 26
+res/res_musiconhold.c | 5
+res/res_timing_dahdi.c | 6
+res/res_timing_kqueue.c | 11
+res/res_timing_pthread.c | 32 -
+res/res_timing_timerfd.c | 45 +
+65 files changed, 2581 insertions(+), 995 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
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@@ -1,0 +1,480 @@
+ Release Summary
+
+ asterisk-11.1.0
+
+ Date: 2012-12-09
+
+ <asteriskteam at digium.com>
+
+ ----------------------------------------------------------------------
+
+ Table of Contents
+
+ 1. Summary
+ 2. Contributors
+ 3. Closed Issues
+ 4. Other Changes
+ 5. Diffstat
+
+ ----------------------------------------------------------------------
+
+ Summary
+
+ [Back to Top]
+
+ This release includes only bug fixes. The changes included were made only
+ to address problems that have been identified in this release series.
+ Users should be able to safely upgrade to this version if this release
+ series is already in use. Users considering upgrading from a previous
+ release series are strongly encouraged to review the UPGRADE.txt document
+ as well as the CHANGES document for information about upgrading to this
+ release series.
+
+ The data in this summary reflects changes that have been made since the
+ previous release, asterisk-11.0.0.
+
+ ----------------------------------------------------------------------
+
+ Contributors
+
+ [Back to Top]
+
+ This table lists the people who have submitted code, those that have
+ tested patches, as well as those that reported issues on the issue tracker
+ that were resolved in this release. For coders, the number is how many of
+ their patches (of any size) were committed into this release. For testers,
+ the number is the number of times their name was listed as assisting with
+ testing a patch. Finally, for reporters, the number is the number of
+ issues that they reported that were closed by commits that went into this
+ release.
+
+ Coders Testers Reporters
+ 6 bebuild 1 Dmitry Burilov 3 coreyfarrell
+ 6 jrose 1 mjordan 3 kmoore
+ 6 rmudgett 1 rmudgett 3 mjordan
+ 5 file 1 Thomas Arimont 2 tomaso
+ 5 mjordan 2 ulogic
+ 4 jcolp 1 chappell
+ 3 Corey Farrell 1 danjenkins
+ 3 kmoore 1 daren
+ 3 mmichelson 1 darius
+ 2 JoshE 1 deniz
+ 2 Richard Miller 1 feyfre
+ 2 sruffell 1 gamegamer43
+ 2 wdoekes 1 gkelleter
+ 1 Bryan Walters 1 jbigelow
+ 1 Daniel O'Connor 1 licedey
+ 1 David Chappell 1 n8ideas
+ 1 elguero 1 netaskd
+ 1 feyfre 1 pciccone
+ 1 Guenther Kelleter 1 sruffell
+ 1 igorg 1 stocksy
+ 1 jbigelow 1 tblancher
+ 1 lathama 1 vilius365
+ 1 twilson
+ 1 tzafrir
+ 1 wedhorn
+
+ ----------------------------------------------------------------------
+
+ Closed Issues
+
+ [Back to Top]
+
+ This is a list of all issues from the issue tracker that were closed by
+ changes that went into this release.
+
+ Category: Applications/app_confbridge
+
+ ASTERISK-19448: ConfBridge crashes Asterisk when no timing module loaded.
+ Revision: 375506
+ Reporter: feyfre
+ Coders: feyfre
+
+ ASTERISK-20601: Confbridge recording does not work
+ Revision: 375471
+ Reporter: vilius365
+ Coders: jrose
+
+ Category: Applications/app_queue
+
+ ASTERISK-20623: App_queue doesn't increment number of busy agent in
+ certain situations
+ Revision: 375469
+ Reporter: gamegamer43
+ Coders: Bryan Walters
+
+ Category: Channels/chan_motif
+
+ ASTERISK-20554: Outgoing calls fail to establish audio due to ICE
+ negotiation failures
+ Revision: 374850
+ Reporter: mjordan
+ Coders: jcolp
+
+ Category: Channels/chan_sip/General
+
+ ASTERISK-18203: Problems with NAT on realtime peers (and maybe static
+ ones)
+ Revision: 375437
+ Reporter: daren
+ Coders: JoshE
+
+ ASTERISK-20487: Failure to have OpenSSL w/ SRTP support results in
+ confusing error message
+ Revision: 374755
+ Reporter: mjordan
+ Coders: jcolp
+
+ ASTERISK-20487: Failure to have OpenSSL w/ SRTP support results in
+ confusing error message
+ Revision: 374756
+ Reporter: mjordan
+ Coders: jcolp
+
+ ASTERISK-20561: Asterisk 1.8 allows the # character in SIP URI, 10 and
+ higher versions do not - need to document in UPGRADE.txt possibly other
+ places?
+ Revision: 375847
+ Reporter: deniz
+ Coders: jrose
+
+ ASTERISK-20572: Realtime Peers behind NAT are Set to RFC1918 private
+ address after sip reload
+ Revision: 375437
+ Reporter: n8ideas
+ Coders: JoshE
+
+ Category: Channels/chan_sip/Messaging
+
+ ASTERISK-20545: chan_sip loads too early because of exposed global symbols
+ Revision: 374842
+ Reporter: kmoore
+ Coders: mmichelson
+
+ Category: Channels/chan_sip/Registration
+
+ ASTERISK-20611: sip registery lost after sip reload
+ Revision: 375575
+ Reporter: licedey
+ Coders: jrose
+
+ Category: Channels/chan_sip/TCP-TLS
+
+ ASTERISK-20212: Deadlock / TCP SIP Stack
+ Revision: 374914
+ Reporter: pciccone
+ Coders: mmichelson
+
+ ASTERISK-20559: SIP TCP/TLS: When checking the CA certificate fails, the
+ call still goes through
+ Revision: 375148
+ Reporter: kmoore
+ Coders: kmoore
+
+ Category: Codecs/codec_gsm
+
+ ASTERISK-20457: GSM encoding is not thread safe
+ Revision: 375288
+ Reporter: ulogic
+ Coders: Richard Miller
+
+ Category: Contrib/General
+
+ ASTERISK-20260: Increase robustness of ast_tls_cert
+ Revision: 375327
+ Reporter: darius
+ Coders: Daniel O'Connor
+
+ Category: Core/AstDB
+
+ ASTERISK-20647: [patch] Failure to cleanup SQLite3 statements during exit
+ causes call to sqlite3_close to fail; leaks memory
+ Revision: 375763
+ Reporter: coreyfarrell
+ Coders: Corey Farrell
+
+ Category: Core/BuildSystem
+
+ ASTERISK-20483: Allow Asterisk to report git SHAs in version string.
+ Revision: 375191
+ Reporter: sruffell
+ Coders: sruffell
+
+ Category: Core/General
+
+ ASTERISK-20648: [patch] - Memory leaks in xmldoc
+ Revision: 375756
+ Reporter: coreyfarrell
+ Testers: mjordan
+ Coders: Corey Farrell
+
+ Category: Core/ManagerInterface
+
+ ASTERISK-20544: action_originate called via ast_hook_send_action causes a
+ segfault
+ Revision: 374792
+ Reporter: kmoore
+ Coders: kmoore
+
+ ASTERISK-20646: [patch] - manager_shutdown fails to completely shutdown
+ AMI and leaks memory
+ Revision: 375795
+ Reporter: coreyfarrell
+ Coders: Corey Farrell
+
+ Category: Core/PBX
+
+ ASTERISK-20455: dialplan fails to run the invalid "i" extension due to an
+ uninitialized variable dat_exten in main/pbx.c
+ Revision: 374771
+ Reporter: ulogic
+ Coders: Richard Miller
+
+ Category: Documentation
+
+ ASTERISK-14435: [patch] Add option and description to
+ chan_dahdi.conf.sample
+ Revision: 374729
+ Reporter: jbigelow
+ Coders: jbigelow, sruffell
+
+ Category: Features/Parking
+
+ ASTERISK-19382: Park() ignores 'r' option, plays default MOH instead.
+ Revision: 375390
+ Reporter: stocksy
+ Coders: rmudgett
+
+ Category: PBX/pbx_realtime
+
+ ASTERISK-18203: Problems with NAT on realtime peers (and maybe static
+ ones)
+ Revision: 375437
+ Reporter: daren
+ Coders: JoshE
+
+ ASTERISK-20572: Realtime Peers behind NAT are Set to RFC1918 private
+ address after sip reload
+ Revision: 375437
+ Reporter: n8ideas
+ Coders: JoshE
+
+ Category: PBX/pbx_spool
+
+ ASTERISK-17231: [patch] unopenable spool files not deleted
+ Revision: 374708
+ Reporter: chappell
+ Coders: David Chappell
+
+ Category: Resources/res_calendar_ews
+
+ ASTERISK-19738: Calendar EWS does not attempt to extract the Body element
+ in a CalendarItem and populate the description event field
+ Revision: 375532
+ Reporter: netaskd
+ Testers: Dmitry Burilov
+ Coders: twilson
+
+ Category: Resources/res_http_websocket
+
+ ASTERISK-20631: Unable to connect via WebRTC
+ Revision: 375559
+ Reporter: danjenkins
+ Coders: jcolp
+
+ ----------------------------------------------------------------------
+
+ Commits Not Associated with an Issue
+
+ [Back to Top]
+
+ This is a list of all changes that went into this release that did not
+ directly close an issue from the issue tracker. The commits may have been
+ marked as being related to an issue. If that is the case, the issue
+ numbers are listed here, as well.
+
+ +------------------------------------------------------------------------+
+ | Revision | Author | Summary | Issues Referenced |
+ |----------+------------+----------------------------+-------------------|
+ | | | Consider the Google Talk | |
+ | 374833 | file | content stanza name | |
+ | | | (jin:content) valid. | |
+ |----------+------------+----------------------------+-------------------|
+ | 374851 | file | Remove code that should | ASTERISK-20554 |
+ | | | not have gotten in. | |
+ |----------+------------+----------------------------+-------------------|
+ | | | Fix a bug where audio on | |
+ | 374877 | file | Google Voice would not | |
+ | | | work due to ignoring | |
+ | | | candidates. | |
+ |----------+------------+----------------------------+-------------------|
+ | 374932 | kmoore | Avoid a segfault on | |
+ | | | invalid format names | |
+ |----------+------------+----------------------------+-------------------|
+ | 374995 | tzafrir | Update config.guess and | |
+ | | | config.sub: 2012-10-10 | |
+ |----------+------------+----------------------------+-------------------|
+ | 375016 | igorg | | |
+ |----------+------------+----------------------------+-------------------|
+ | 375027 | mmichelson | Fix some potential misuses | |
+ | | | of ast_str in the code. | |
+ |----------+------------+----------------------------+-------------------|
+ | | | Remove a log message that | |
+ | 375051 | file | was left in accidentally | |
+ | | | from call-id logging | |
+ | | | development. | |
+ |----------+------------+----------------------------+-------------------|
+ | | | Update sip_request_call | |
+ | 375079 | wdoekes | SIP dial string | |
+ | | | documentation. | |
+ |----------+------------+----------------------------+-------------------|
+ | 375113 | wdoekes | Fixes to the fd-oriented | |
+ | | | SIP TCP reads. | |
+ |----------+------------+----------------------------+-------------------|
+ | | | app_queue: Make ordering | |
+ | 375219 | jrose | of rrmemory/rrordered | AST-989 |
+ | | | persist over add/remove | |
+ | | | members | |
+ |----------+------------+----------------------------+-------------------|
+ | 375247 | jrose | app_queue: add upgrade | AST-989 |
+ | | | notes for 375216 | |
+ |----------+------------+----------------------------+-------------------|
+ | 375486 | jrose | mixmonitor: Add a test | |
+ | | | event | |
+ |----------+------------+----------------------------+-------------------|
+ | | | Fix Wrong Result In Debug | |
+ | 375613 | elguero | Message For SDP Origin | |
+ | | | Processing | |
+ |----------+------------+----------------------------+-------------------|
+ | 375627 | rmudgett | Multiple revisions | |
+ | | | 375519-375524 | |
+ |----------+------------+----------------------------+-------------------|
+ | 375660 | wedhorn | Fix for chan_skinny | |
+ | | | leaving RTP ports open | |
+ |----------+------------+----------------------------+-------------------|
+ | 375661 | rmudgett | Things don't need to be | |
+ | | | that const. | |
+ |----------+------------+----------------------------+-------------------|
+ | 375702 | lathama | Doxygen Updates | ASTERISK-20259 |
[... 134 lines stripped ...]
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