[asterisk-commits] bebuild: tag 10.1.0-rc1 r349398 - /tags/10.1.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Dec 29 13:41:41 CST 2011
Author: bebuild
Date: Thu Dec 29 13:41:37 2011
New Revision: 349398
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349398
Log:
Importing release summary for 10.1.0-rc1 release.
Added:
tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.html (with props)
tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.txt (with props)
Added: tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.html
URL: http://svnview.digium.com/svn/asterisk/tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.html?view=auto&rev=349398
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--- tags/10.1.0-rc1/asterisk-10.1.0-rc1-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-10.1.0-rc1</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-10.1.0-rc1</h3>
+<h3 align="center">Date: 2011-12-29</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="#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-10.0.0.</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>
+33 rmudgett<br/>
+12 wdoekes<br/>
+11 jrose<br/>
+10 mjordan<br/>
+10 twilson<br/>
+8 kmoore<br/>
+3 kpfleming<br/>
+3 may<br/>
+3 mnicholson<br/>
+3 seanbright<br/>
+2 dvossel<br/>
+2 lmadsen<br/>
+2 pabelanger<br/>
+2 schmidts<br/>
+2 tilghman<br/>
+1 irroot<br/>
+</td>
+<td>
+</td>
+<td>
+</td>
+</tr>
+</table>
+<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/10?view=revision&revision=344004">344004</a></td><td>rmudgett</td><td>Residual changes for Asterisk v10 branch from ASTERISK-18747.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18747">ASTERISK-18747</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344049">344049</a></td><td>mnicholson</td><td>don't call ltohl() twice on the same value</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18739">ASTERISK-18739</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344103">344103</a></td><td>kmoore</td><td>Fix pin parameter behavior regression in MeetMe</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18488">ASTERISK-18488</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344159">344159</a></td><td>may</td><td>Generate response to Status Enquiry message with Status q.931 message.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18748">ASTERISK-18748</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344160">344160</a></td><td>may</td><td>delete svn:mergeinfo</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344175">344175</a></td><td>twilson</td><td>Add a unit test for ast_sockaddr_split_hostport</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344216">344216</a></td><td>twilson</td><td>Don't treat a host:port string as a domain</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344271">344271</a></td><td>rmudgett</td><td>Fix deadlock during dialplan reload.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18740">ASTERISK-18740</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344334">344334</a></td><td>mnicholson</td><td>only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18490">ASTERISK-18490</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344386">344386</a></td><td>kmoore</td><td>Fix several bugs with SDP parsing and well-formedness of responses</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16558">ASTERISK-16558</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344440">344440</a></td><td>kmoore</td><td>Fix another incorrect case with meetme's PIN logic and add documentation</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344493">344493</a></td><td>dvossel</td><td>Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18829">ASTERISK-18829</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344537">344537</a></td><td>rmudgett</td><td>Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18152">ASTERISK-18152</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344540">344540</a></td><td>rmudgett</td><td>Fix potential deadlock calling ast_call() with channel locks held.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344557">344557</a></td><td>rmudgett</td><td>Fix app_macro.c MODULEINFO section termination.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18848">ASTERISK-18848</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344609">344609</a></td><td>jrose</td><td>Fix a segmentation fault when using an extension with CID matching and no CID.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18392">ASTERISK-18392</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344662">344662</a></td><td>rmudgett</td><td>Make CLI "core show channel" not hold the channel lock during console output.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18571">ASTERISK-18571</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344716">344716</a></td><td>rmudgett</td><td>Check sip.conf maxforwards parameter for range 1 <= x <= 255.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344770">344770</a></td><td>kmoore</td><td>Fix regression introduced by SDP fixups</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344836">344836</a></td><td>wdoekes</td><td>Fix bad quoting of multiline mxml opaque_data that caused invalid xml.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18852">ASTERISK-18852</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344839">344839</a></td><td>wdoekes</td><td>Remove unneeded if(params) checks in reqresp_parser.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344842">344842</a></td><td>mjordan</td><td>Video format was treated as audio when removed from the file playback scheduler</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18682">ASTERISK-18682</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344845">344845</a></td><td>wdoekes</td><td>Use __alignof__ instead of sizeof for stringfield length storage.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344900">344900</a></td><td>twilson</td><td>Don't forget to rescan MOH files for cached realtime classes</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18039">ASTERISK-18039</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=344966">344966</a></td><td>irroot</td><td>mISDN Round Robin break when no channel is available</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345064">345064</a></td><td>kmoore</td><td>Ensure that a null vmexten does not cause a segfault</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345117">345117</a></td><td>jrose</td><td>Moves voicemail setup password entry to the end of the setup process.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18282">ASTERISK-18282</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345161">345161</a></td><td>wdoekes</td><td>Update reqresp_parser parse_uri doxygen comments.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18572">ASTERISK-18572</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345164">345164</a></td><td>twilson</td><td>Don't read past end of input when calling write()</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345220">345220</a></td><td>rmudgett</td><td>Fix Progress spelling error in main/pbx.c.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18857">ASTERISK-18857</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345275">345275</a></td><td>rmudgett</td><td>Restore SIP DTMF overlap dialing method.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17288">ASTERISK-17288</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18702">ASTERISK-18702</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345290">345290</a></td><td>rmudgett</td><td>Make queue log indicate if ADDMEMBER is paused for AMI and realtime.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18645">ASTERISK-18645</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345371">345371</a></td><td>rmudgett</td><td>Fix typo in sig_pri using wrong structure name.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18868">ASTERISK-18868</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345432">345432</a></td><td>rmudgett</td><td>Make FastAGI HANGUP show up in AGI debug output.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18723">ASTERISK-18723</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345488">345488</a></td><td>jrose</td><td>Guarantee messages go into the right folders with multiple recipients</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18245">ASTERISK-18245</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18246">ASTERISK-18246</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345558">345558</a></td><td>rmudgett</td><td>Remove dead code since pri_grab() can never fail.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345640">345640</a></td><td>tilghman</td><td>Fix a change in behavior in 'database show' from 1.8.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18886">ASTERISK-18886</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345683">345683</a></td><td>tilghman</td><td>Update the documentation to better clarify how the existing commands work.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345830">345830</a></td><td>twilson</td><td>Default to nat=yes; warn when nat in general and peer differ</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18862">ASTERISK-18862</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345882">345882</a></td><td>pabelanger</td><td>Add missing sound_only_one config variable</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18895">ASTERISK-18895</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345924">345924</a></td><td>wdoekes</td><td>Clarify why the AST_LOG_* macros exist next to the LOG_* macros.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17973">ASTERISK-17973</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=345977">345977</a></td><td>rmudgett</td><td>Fix dnsmgr entries to ask for the same address family each time.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346029">346029</a></td><td>pabelanger</td><td>Added support level for new modules</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346031">346031</a></td><td>twilson</td><td>Resume playing existing hold music for cached realtime MOH</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18039">ASTERISK-18039</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18912">ASTERISK-18912</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346040">346040</a></td><td>mjordan</td><td>Fixed SendMessage stripping extension from To: header in SIP MESSAGE</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18903">ASTERISK-18903</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346087">346087</a></td><td>kmoore</td><td>Fix res_jabber resource leaks</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346145">346145</a></td><td>wdoekes</td><td>Fix ast_str_truncate signedness warning and documentation.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346198">346198</a></td><td>wdoekes</td><td>Minor cleanup in chan_sip get_msg_text() function.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346240">346240</a></td><td>rmudgett</td><td>Fix calls to ast_get_ip() not initializing the address family.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346293">346293</a></td><td>schmidts</td><td>Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18693">ASTERISK-18693</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346349">346349</a></td><td>dvossel</td><td>Fixes memory leak in message API.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346473">346473</a></td><td>lmadsen</td><td>Update queues.conf.sample documentation.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17413">ASTERISK-17413</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346565">346565</a></td><td>jrose</td><td>r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18700">ASTERISK-18700</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18345">ASTERISK-18345</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18342">ASTERISK-18342</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346698">346698</a></td><td>jrose</td><td>Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18925">ASTERISK-18925</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346701">346701</a></td><td>rmudgett</td><td>Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18327">ASTERISK-18327</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346763">346763</a></td><td>may</td><td>process null frame pointer returned by ast_rtp_instance_read correctly</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16697">ASTERISK-16697</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346856">346856</a></td><td>mjordan</td><td>Update SIP MESSAGE To parsing to correctly handle URI</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18903">ASTERISK-18903</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346900">346900</a></td><td>wdoekes</td><td>For SIP REGISTER fix domain-only URIs and domain ACL bypass.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18389">ASTERISK-18389</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18741">ASTERISK-18741</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346952">346952</a></td><td>kmoore</td><td>Fix chan_jingle/gtalk load regression introduced in r346087</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=346955">346955</a></td><td>jrose</td><td>Resolve duplicate label used in multiple priorities for the same extension.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18807">ASTERISK-18807</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347007">347007</a></td><td>rmudgett</td><td>Restore call progress code for analog ports.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18841">ASTERISK-18841</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347068">347068</a></td><td>mjordan</td><td>Fixed crash from orphaned MWI subscriptions in chan_sip</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18663">ASTERISK-18663</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347124">347124</a></td><td>wdoekes</td><td>Move setting of voicemail zonetag and locale up a bit.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18838">ASTERISK-18838</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347146">347146</a></td><td>wdoekes</td><td>Add regression tests for issue ASTERISK-18838.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347167">347167</a></td><td>wdoekes</td><td>Don't allow transport=tcp when tcpenable=no.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18837">ASTERISK-18837</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347240">347240</a></td><td>jrose</td><td>Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18804">ASTERISK-18804</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347293">347293</a></td><td>rmudgett</td><td>Make SIP INFO messages for dtmf-relay signals case insensitive.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18924">ASTERISK-18924</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347344">347344</a></td><td>twilson</td><td>Add ASTSBINDIR to the list of configurable paths</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18959">ASTERISK-18959</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347383">347383</a></td><td>jrose</td><td>Fix: Meetme recording variables from realtime DB use null entries over channel variables</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18873">ASTERISK-18873</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347439">347439</a></td><td>rmudgett</td><td>Update AMI Getvar and Setvar documentation about supplying a channel name.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18958">ASTERISK-18958</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347532">347532</a></td><td>twilson</td><td>Don't crash on INFO automon request with no channel</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18805">ASTERISK-18805</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347600">347600</a></td><td>rmudgett</td><td>Mark channel running the h exten with the soft-hangup flag.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18811">ASTERISK-18811</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347656">347656</a></td><td>jrose</td><td>Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347727">347727</a></td><td>wdoekes</td><td>Fix regression when using tcpenable=no and tlsenable=yes.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347812">347812</a></td><td>rmudgett</td><td>Fix some parsing issues in add_exten_to_pattern_tree().</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18909">ASTERISK-18909</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347953">347953</a></td><td>rmudgett</td><td>Update sample configs to put incoming calls into context public.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-14122">ASTERISK-14122</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347955">347955</a></td><td>rmudgett</td><td>Reverting -r347953 for ASTERISK-14122</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=347996">347996</a></td><td>twilson</td><td>Add a separate buffer for SRTCP packets</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18889">ASTERISK-18889</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348056">348056</a></td><td>schmidts</td><td>Fix possible misshandling of an incoming SIP response as a peer poke response.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18940">ASTERISK-18940</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348102">348102</a></td><td>rmudgett</td><td>Fix FollowMe CallerID on outgoing calls.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17557">ASTERISK-17557</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348155">348155</a></td><td>jrose</td><td>Document PARKINGSLOT variable in features.conf.sample</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16239">ASTERISK-16239</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348158">348158</a></td><td>jrose</td><td>Fix accidental use of tabs instead of spaces from previous features.conf.sample change</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348211">348211</a></td><td>mjordan</td><td>Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348213">348213</a></td><td>mnicholson</td><td>Don't clear LOCALSTATIONID before sending or receiving. The user may set that</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18921">ASTERISK-18921</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348265">348265</a></td><td>mjordan</td><td>Added support for all slin formats to app_originate</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348311">348311</a></td><td>rmudgett</td><td>Fix ParkAndAnnounce to pass the CallerID to the announcing channel.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348363">348363</a></td><td>rmudgett</td><td>Fix crash during CDR update.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18836">ASTERISK-18836</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348405">348405</a></td><td>rmudgett</td><td>Fix cut and past error in ast_call_forward().</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18836">ASTERISK-18836</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348465">348465</a></td><td>rmudgett</td><td>Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348517">348517</a></td><td>kpfleming</td><td>Correct two flaws in sip.conf.sample related to AST-2011-013.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348605">348605</a></td><td>lmadsen</td><td>Update documentation for MESSAGE_SEND_STATUS variable.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19056">ASTERISK-19056</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348648">348648</a></td><td>rmudgett</td><td>Fix crashes on other platforms caused by interference from Darwin weak symbol support.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18728">ASTERISK-18728</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348736">348736</a></td><td>rmudgett</td><td>Fix chan_iax2 to not report an RDNIS number if it is blank.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17152">ASTERISK-17152</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348790">348790</a></td><td>rmudgett</td><td>Make apps/confbridge ignore *.i files also.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348793">348793</a></td><td>rmudgett</td><td>Make codecs/speex ignore *.i files also.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348845">348845</a></td><td>twilson</td><td>Allow packetization vaules > 127</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18876">ASTERISK-18876</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348846">348846</a></td><td>mjordan</td><td>Add Asterisk TestSuite event hooks to support ConfBridge testing</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19059">ASTERISK-19059</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348889">348889</a></td><td>mjordan</td><td>Fix for memory leaks / cleanup in cel_pgsql</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18879">ASTERISK-18879</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348952">348952</a></td><td>rmudgett</td><td>Fix extension state callback references in chan_sip.</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17760">ASTERISK-17760</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18844">ASTERISK-18844</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=348993">348993</a></td><td>kmoore</td><td>Fix missing doc tags found while fixing ASTERISK-18689</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18689">ASTERISK-18689</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349045">349045</a></td><td>seanbright</td><td>In ChanSpy, don't create audiohooks that will never be used.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349145">349145</a></td><td>seanbright</td><td>Once an audiohook is attached to a channel, we continue to transcode all of the</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349195">349195</a></td><td>mjordan</td><td>Fix timing source dependency issues with MOH</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17474">ASTERISK-17474</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349248">349248</a></td><td>kpfleming</td><td>Improve T.38 gateway V.21 preamble detection.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349250">349250</a></td><td>kpfleming</td><td>Tell Subversion to gnore the 'astdb2bdb' binary file if it exists.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349290">349290</a></td><td>seanbright</td><td>Use ast_audiohook_write_list_empty to determine if our lists are empty instead</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10?view=revision&revision=349340">349340</a></td><td>mjordan</td><td>Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19040">ASTERISK-19040</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19128">ASTERISK-19128</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-17725">ASTERISK-17725</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-18340">ASTERISK-18340</a>, <a href="https://issues.asterisk.org/jira/browse/ASTERISK-19095">ASTERISK-19095</a></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>
+CHANGES | 5
+Makefile | 4
+UPGRADE-1.8.txt | 23
+addons/chan_ooh323.c | 2
+addons/ooh323c/src/oochannels.c | 4
+addons/ooh323c/src/ooh245.c | 17
+addons/ooh323c/src/ooh323.c | 1
+addons/ooh323c/src/ooq931.c | 179 ++++++
+addons/ooh323c/src/ooq931.h | 8
+addons/ooh323c/src/ootypes.h | 3
+apps/app_authenticate.c | 15
+apps/app_chanspy.c | 56 +-
+apps/app_confbridge.c | 6
+apps/app_dial.c | 2
+apps/app_followme.c | 201 +++----
+apps/app_macro.c | 2
+apps/app_meetme.c | 34 -
+apps/app_originate.c | 8
+apps/app_parkandannounce.c | 19
+apps/app_queue.c | 187 ++++---
+apps/app_voicemail.c | 329 ++++++++----
+apps/confbridge/conf_config_parser.c | 2
+bridges/bridge_builtin_features.c | 13
+build_tools/make_defaults_h | 1
+cel/cel_pgsql.c | 37 -
+channels/chan_dahdi.c | 12
+channels/chan_gtalk.c | 25
+channels/chan_h323.c | 3
+channels/chan_iax2.c | 10
+channels/chan_jingle.c | 46 +
+channels/chan_misdn.c | 16
+channels/chan_sip.c | 907 ++++++++++++++++++++++------------
+channels/chan_skinny.c | 1
+channels/sig_analog.c | 13
+channels/sig_analog.h | 1
+channels/sig_pri.c | 175 ++----
+channels/sip/include/reqresp_parser.h | 14
+channels/sip/include/sip.h | 82 +--
+channels/sip/reqresp_parser.c | 198 +++----
+configs/asterisk.conf.sample | 1
+configs/features.conf.sample | 2
+configs/queues.conf.sample | 9
+configs/res_stun_monitor.conf.sample | 17
+configs/sip.conf.sample | 26
+configure.ac | 34 +
+formats/format_wav.c | 6
+funcs/func_cdr.c | 20
+include/asterisk/acl.h | 25
+include/asterisk/cdr.h | 32 -
+include/asterisk/dnsmgr.h | 19
+include/asterisk/dsp.h | 5
+include/asterisk/format_pref.h | 2
+include/asterisk/jabber.h | 5
+include/asterisk/logger.h | 4
+include/asterisk/message.h | 3
+include/asterisk/module.h | 1
+include/asterisk/paths.h | 1
+include/asterisk/pbx.h | 40 +
+include/asterisk/res_fax.h | 4
+include/asterisk/stringfields.h | 7
+include/asterisk/strings.h | 10
+include/asterisk/stun.h | 43 +
+include/asterisk/tcptls.h | 7
+include/asterisk/utils.h | 63 +-
+main/acl.c | 12
+main/app.c | 3
+main/asterisk.c | 18
+main/audiohook.c | 4
+main/bridging.c | 25
+main/channel.c | 128 +++-
+main/cli.c | 32 -
+main/db.c | 36 -
+main/dnsmgr.c | 18
+main/dsp.c | 147 -----
+main/features.c | 28 -
+main/file.c | 57 +-
+main/manager.c | 15
+main/message.c | 12
+main/pbx.c | 515 ++++++++++++-------
+main/rtp_engine.c | 8
+main/say.c | 2
+main/stun.c | 126 ++--
+main/tcptls.c | 55 +-
+main/utils.c | 18
+res/res_agi.c | 4
+res/res_fax.c | 195 ++++---
+res/res_fax_spandsp.c | 85 +++
+res/res_format_attr_celt.c | 4
+res/res_format_attr_silk.c | 4
+res/res_jabber.c | 198 ++++---
+res/res_jabber.exports.in | 2
+res/res_monitor.c | 6
+res/res_musiconhold.c | 38 +
+res/res_rtp_asterisk.c | 43 +
+res/res_srtp.c | 10
+res/res_stun_monitor.c | 302 +++++++----
+res/res_timing_dahdi.c | 2
+res/res_timing_pthread.c | 2
+res/res_timing_timerfd.c | 2
+tests/test_netsock2.c | 71 ++
+100 files changed, 3445 insertions(+), 1829 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
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--- tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.txt (added)
+++ tags/10.1.0-rc1/asterisk-10.1.0-rc1-summary.txt Thu Dec 29 13:41:37 2011
@@ -1,0 +1,553 @@
+ Release Summary
+
+ asterisk-10.1.0-rc1
+
+ Date: 2011-12-29
+
+ <asteriskteam at digium.com>
+
+ ----------------------------------------------------------------------
+
+ Table of Contents
+
+ 1. Summary
+ 2. Contributors
+ 3. Other Changes
+ 4. 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
[... 540 lines stripped ...]
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