[svn-commits] lmadsen: tag 1.6.0.20 r235512 - /tags/1.6.0.20/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Dec 17 14:19:22 CST 2009
Author: lmadsen
Date: Thu Dec 17 14:19:20 2009
New Revision: 235512
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=235512
Log:
Importing release summary for 1.6.0.20 release.
Added:
tags/1.6.0.20/asterisk-1.6.0.20-summary.html (with props)
tags/1.6.0.20/asterisk-1.6.0.20-summary.txt (with props)
Added: tags/1.6.0.20/asterisk-1.6.0.20-summary.html
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.20/asterisk-1.6.0.20-summary.html?view=auto&rev=235512
==============================================================================
--- tags/1.6.0.20/asterisk-1.6.0.20-summary.html (added)
+++ tags/1.6.0.20/asterisk-1.6.0.20-summary.html Thu Dec 17 14:19:20 2009
@@ -1,0 +1,440 @@
+<!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-1.6.0.20</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-1.6.0.20</h3>
+<h3 align="center">Date: 2009-12-17</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-1.6.0.18.</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>
+13 tilghman<br/>
+10 dvossel<br/>
+7 russell<br/>
+4 kpfleming<br/>
+4 mnicholson<br/>
+3 alecdavis<br/>
+2 atis<br/>
+2 dimas<br/>
+2 file<br/>
+2 jpeeler<br/>
+2 mnick<br/>
+2 oej<br/>
+1 bklang<br/>
+1 diruggles<br/>
+1 jsmith<br/>
+1 Laureano<br/>
+1 lmadsen<br/>
+1 qwell<br/>
+1 vnovy<br/>
+</td>
+<td>
+5 dvossel<br/>
+5 falves11<br/>
+3 alecdavis<br/>
+3 mnicholson<br/>
+3 parisioa<br/>
+3 tilghman<br/>
+2 mnick<br/>
+1 abelbeck<br/>
+1 ajohnson<br/>
+1 AlexMS<br/>
+1 amorsen<br/>
+1 diLLec<br/>
+1 Laureano<br/>
+1 lmadsen<br/>
+1 oej<br/>
+1 OrNix<br/>
+1 Parantido<br/>
+1 qwell<br/>
+1 rmudgett<br/>
+1 russell<br/>
+1 thedavidfactor<br/>
+1 zalex1953<br/>
+</td>
+<td>
+3 falves11<br/>
+2 alecdavis<br/>
+2 atis<br/>
+2 dimas<br/>
+2 parisioa<br/>
+2 wdoekes<br/>
+1 abelbeck<br/>
+1 ajohnson<br/>
+1 AlexMS<br/>
+1 amorsen<br/>
+1 bklang<br/>
+1 CGMChris<br/>
+1 corruptor<br/>
+1 dant<br/>
+1 davidw<br/>
+1 dcabot<br/>
+1 diLLec<br/>
+1 jiddings<br/>
+1 mgernoth<br/>
+1 msetim<br/>
+1 nicchap<br/>
+1 OrNix<br/>
+1 Parantido<br/>
+1 patrol-cz<br/>
+1 rmudgett<br/>
+1 sgimeno<br/>
+1 Shagg63<br/>
+1 udosw<br/>
+1 vnovy<br/>
+1 zalex1953<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_amd</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16239">#16239</a>: AMD() incorrectly sets AMDCAUSE channel variable<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232357">232357</a><br/>
+Reporter: CGMChris<br/>
+Coders: file<br/>
+<br/>
+<h3>Category: Applications/app_dial</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16337">#16337</a>: [patch] Segmentation Fault on Originate command.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232272">232272</a><br/>
+Reporter: Parantido<br/>
+Testers: Parantido, dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Applications/app_directory</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16437">#16437</a>: [patch] Unable to escape back to dialplan or operator, using 'o' and 'a' extensions in dialcontext<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234894">234894</a><br/>
+Reporter: alecdavis<br/>
+Testers: alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Applications/app_externalivr</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16305">#16305</a>: [patch] ExternalIVR confuses AGI by double closing FDs<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232811">232811</a><br/>
+Reporter: diLLec<br/>
+Testers: thedavidfactor, diLLec<br/>
+Coders: diruggles<br/>
+<br/>
+<h3>Category: Applications/app_meetme</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16247">#16247</a>: [patch] Muted user remains talking forever<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234404">234404</a><br/>
+Reporter: dimas<br/>
+Coders: dimas<br/>
+<br/>
+<h3>Category: Applications/app_mixmonitor</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16152">#16152</a>: [patch] MixMonitor thread doesn't exit until channel is dropped<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230512">230512</a><br/>
+Reporter: AlexMS<br/>
+Testers: dvossel, AlexMS<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Applications/app_voicemail</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15625">#15625</a>: [patch] Prepending to a voicemail on forward causes locked sip channel and large file filling disk space<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231691">231691</a><br/>
+Reporter: Shagg63<br/>
+Testers: mnicholson<br/>
+Coders: mnicholson<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16291">#16291</a>: app_voicemail.c strip_control() strips more than just control chars<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233167">233167</a><br/>
+Reporter: wdoekes<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Channels/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16058">#16058</a>: [patch] Crash in local_ast_moh_start / ast_indicate_data due to AST_CONTROL_HOLD with bad pointer<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231928">231928</a><br/>
+Reporter: atis<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16242">#16242</a>: [patch] Comfort noise frame with f->data NULL but f->datalen 160<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231517">231517</a><br/>
+Reporter: amorsen<br/>
+Testers: amorsen, oej, dvossel<br/>
+Coders: oej<br/>
+<br/>
+<h3>Category: Channels/chan_agent</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14590">#14590</a>: [patch] updatecdr='yes' in agents.conf is not working<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230631">230631</a><br/>
+Reporter: msetim<br/>
+Testers: Laureano, mnicholson<br/>
+Coders: Laureano<br/>
+<br/>
+<h3>Category: Channels/chan_dahdi</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15972">#15972</a>: [patch] Dropping frame since I'm still dialing on Zap/... (resp. DAHDi/...) with DIGITAL calls<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232092">232092</a><br/>
+Reporter: udosw<br/>
+Testers: alecdavis<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Channels/chan_iax2</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16094">#16094</a>: [patch] iax2 show cache, locks channels.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230729">230729</a><br/>
+Reporter: alecdavis<br/>
+Testers: alecdavis, dvossel<br/>
+Coders: alecdavis<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15356">#15356</a>: After a few thousand calls, or at random, Asterisk stops receiving events from the network<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234131">234131</a><br/>
+Reporter: falves11<br/>
+Testers: falves11<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=15716">#15716</a>: [patch] chan_sip fails to destroy channels in INVITE when no response received<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234131">234131</a><br/>
+Reporter: dant<br/>
+Testers: falves11<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16268">#16268</a>: [patch] Last line of SDP is not being parsed<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230782">230782</a><br/>
+Reporter: sgimeno<br/>
+Coders: kpfleming<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16186">#16186</a>: [patch] T.38 reinvite fails after receiving "415 Unsupported media type" when it could continue in audio mode<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232346">232346</a><br/>
+Reporter: atis<br/>
+Coders: atis<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Registration</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16298">#16298</a>: [patch] After upgrading to asterisk 1.4.27 Optipoint SIP phone can no longer register<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233475">233475</a><br/>
+Reporter: mgernoth<br/>
+Testers: dvossel<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Channels/chan_sip/T.38</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16387">#16387</a>: [patch] Missing session level connection data (c=) breaks process_sdp()<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233397">233397</a><br/>
+Reporter: zalex1953<br/>
+Testers: mnicholson, zalex1953<br/>
+Coders: mnicholson<br/>
+<br/>
+<h3>Category: Core/BuildSystem</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=14737">#14737</a>: [patch] Cannot flavour (flavor) version number because make_version_c looks in wrong directory<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234701">234701</a><br/>
+Reporter: davidw<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16296">#16296</a>: [patch] menuselect.makeopts: does not properly unselect an option with a leading - (minus)<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234257">234257</a><br/>
+Reporter: abelbeck<br/>
+Testers: abelbeck, qwell, lmadsen<br/>
+Coders: qwell<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15769">#15769</a>: [patch] useless message pops hundreds of times per minute<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233047">233047</a><br/>
+Reporter: falves11<br/>
+Testers: mnick, falves11<br/>
+Coders: mnick<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16106">#16106</a>: [patch] Hangup extension executed twice in 1.6.2 RC2<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231096">231096</a><br/>
+Reporter: ajohnson<br/>
+Testers: ajohnson<br/>
+Coders: jpeeler<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16290">#16290</a>: ast_ouraddrfor doesn't do htons() on the port<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232354">232354</a><br/>
+Reporter: wdoekes<br/>
+Coders: dvossel<br/>
+<br/>
+<h3>Category: Core/ManagerInterface</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16264">#16264</a>: [patch] UserEvent manager action is not ACKed<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232583">232583</a><br/>
+Reporter: dimas<br/>
+Coders: dimas<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16275">#16275</a>: [patch] response to "Action: Events" is not finished by empty line<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232577">232577</a><br/>
+Reporter: vnovy<br/>
+Coders: vnovy<br/>
+<br/>
+<h3>Category: Core/Netsock</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=15627">#15627</a>: [patch] Asterisk runs out of sockets<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234131">234131</a><br/>
+Reporter: falves11<br/>
+Testers: falves11<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16270">#16270</a>: [patch] Asterisk doesn't free udp ports<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234131">234131</a><br/>
+Reporter: corruptor<br/>
+Testers: falves11<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: Documentation</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16223">#16223</a>: [patch] "requirecalltoken" config directive not respected globally<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233284">233284</a><br/>
+Reporter: bklang<br/>
+Coders: bklang<br/>
+<br/>
+<h3>Category: Formats/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16412">#16412</a>: Ignoring unknown format wav & wav49...<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233841">233841</a><br/>
+Reporter: jiddings<br/>
+Testers: russell<br/>
+Coders: russell<br/>
+<br/>
+<h3>Category: General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16113">#16113</a>: [patch] Auto-fallthrough when attempting to enter DTMF using Background() in a Macro()<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=235423">235423</a><br/>
+Reporter: OrNix<br/>
+Testers: OrNix<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16272">#16272</a>: [patch] Language code collisions for certan languages<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232864">232864</a><br/>
+Reporter: patrol-cz<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16446">#16446</a>: segfault error 4<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=235054">235054</a><br/>
+Reporter: nicchap<br/>
+Coders: tilghman<br/>
+<br/>
+<h3>Category: PBX/General</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16166">#16166</a>: [patch] Setting dialplan hint and using a global variable gives incorrect warning.<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233236">233236</a><br/>
+Reporter: rmudgett<br/>
+Testers: mnick, rmudgett<br/>
+Coders: mnick<br/>
+<br/>
+<h3>Category: Resources/res_musiconhold</h3><br/>
+<a href="https://issues.asterisk.org/view.php?id=16207">#16207</a>: [patch] asterisk keeps starting new processes for streaming audio MOH<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232699">232699</a><br/>
+Reporter: dcabot<br/>
+Testers: parisioa, tilghman<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16279">#16279</a>: [patch] asterisk reload causes mpg123 streams to be recreated<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232699">232699</a><br/>
+Reporter: parisioa<br/>
+Testers: parisioa, tilghman<br/>
+Coders: tilghman<br/>
+<br/>
+<a href="https://issues.asterisk.org/view.php?id=16388">#16388</a>: [patch] New music on hold patches cause asterisk + full system hard lock<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233729">233729</a><br/>
+Reporter: parisioa<br/>
+Testers: parisioa, tilghman<br/>
+Coders: tilghman<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/1.6.0?view=revision&revision=230587">230587</a></td><td>dvossel</td><td>audiohook signal trigger on every status change</td>
+<td><a href="https://issues.asterisk.org/view.php?id=14618">#14618</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230878">230878</a></td><td>kpfleming</td><td>Correct fix for issue #16268... the reporter's original patch was very close to correct.</td>
+<td><a href="https://issues.asterisk.org/view.php?id=16268">#16268</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=230882">230882</a></td><td>file</td><td>Change fax detection in chan_sip so it behaves as one would expect.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231192">231192</a></td><td>mnicholson</td><td>Load pbx_lua with global symbols to allow linking with other lua libraries.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231300">231300</a></td><td>tilghman</td><td>After a frame duplication failure, unlock the channel before returning.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231560">231560</a></td><td>dvossel</td><td>app_queue crashes randomly, often during call-transfers</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231693">231693</a></td><td>kpfleming</td><td>Another round of UDPTL stack fixes/improvements:</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231744">231744</a></td><td>mnicholson</td><td>Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=231879">231879</a></td><td>dvossel</td><td>WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232009">232009</a></td><td>russell</td><td>Fix a warning pointed out by buildbot.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=232013">232013</a></td><td>russell</td><td>Fix a build error on FreeBSD.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233112">233112</a></td><td>russell</td><td>Only do frame payload check for HOLD frames.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233230">233230</a></td><td>russell</td><td>unblock a rev.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233614">233614</a></td><td>dvossel</td><td>fixes incorrect logic in ast_uri_encode</td>
+<td><a href="https://issues.asterisk.org/view.php?id=16299">#16299</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233617">233617</a></td><td>atis</td><td>Fix compatibility with valgrind 3.3 and older.</td>
+<td><a href="https://issues.asterisk.org/view.php?id=16388">#16388</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=233881">233881</a></td><td>russell</td><td>Fix breakage of the "module load <module>" CLI command.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234009">234009</a></td><td>russell</td><td>Fix up the faxdetect entry in CHANGES.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234211">234211</a></td><td>tilghman</td><td>Missed a case that emits a WARNING where none is warranted.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234528">234528</a></td><td>oej</td><td>Stop sending 183's after call hangup.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=234636">234636</a></td><td>lmadsen</td><td>Update IMAP build documentation.</td>
+<td><a href="https://issues.asterisk.org/view.php?id=16433">#16433</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=235011">235011</a></td><td>kpfleming</td><td>spandsp does in fact support V.17 modulation at 14.4 kilobits per second,</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=235136">235136</a></td><td>dvossel</td><td>reverses minor sip registration regression</td>
+<td><a href="https://issues.asterisk.org/view.php?id=15539">#15539</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.6.0?view=revision&revision=235332">235332</a></td><td>jsmith</td><td>Add a line showing that we can use CIDR notation.</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>
+CHANGES | 4
+Makefile | 13 -
+apps/app_amd.c | 1
+apps/app_directory.c | 4
+apps/app_externalivr.c | 115 +++++-----
+apps/app_fax.c | 2
+apps/app_meetme.c | 58 +++--
+apps/app_mixmonitor.c | 94 ++++++--
+apps/app_queue.c | 15 +
+apps/app_voicemail.c | 456 ++++++++++++++++++++++++++++---------------
+build_tools/make_version_c | 5
+build_tools/make_version_h | 13 +
+channels/chan_dahdi.c | 6
+channels/chan_iax2.c | 2
+channels/chan_sip.c | 180 +++++++++++-----
+configs/iax.conf.sample | 3
+configs/sip.conf.sample | 6
+contrib/valgrind.supp | 12 -
+doc/tex/imapstorage.tex | 14 +
+formats/format_g723.c | 5
+formats/format_g726.c | 5
+formats/format_g729.c | 5
+formats/format_gsm.c | 5
+formats/format_h263.c | 5
+formats/format_h264.c | 5
+formats/format_ilbc.c | 5
+formats/format_jpeg.c | 5
+formats/format_ogg_vorbis.c | 5
+formats/format_pcm.c | 5
+formats/format_sln.c | 5
+formats/format_sln16.c | 5
+formats/format_vox.c | 5
+formats/format_wav.c | 5
+formats/format_wav_gsm.c | 5
+funcs/func_groupcount.c | 4
+funcs/func_lock.c | 2
+include/asterisk/audiohook.h | 6
+include/asterisk/file.h | 11 +
+include/asterisk/module.h | 8
+include/asterisk/udptl.h | 19 +
+main/acl.c | 2
+main/app.c | 12 -
+main/audiohook.c | 51 ++--
+main/channel.c | 12 +
+main/dsp.c | 7
+main/features.c | 10
+main/file.c | 73 ++++++
+main/loader.c | 123 ++++++-----
+main/manager.c | 5
+main/pbx.c | 18 +
+main/rtp.c | 1
+main/udptl.c | 307 +++++++++++++++++-----------
+main/utils.c | 2
+pbx/pbx_config.c | 29 ++
+pbx/pbx_lua.c | 2
+res/res_agi.c | 3
+res/res_musiconhold.c | 162 +++++++++++----
+57 files changed, 1316 insertions(+), 631 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
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--- tags/1.6.0.20/asterisk-1.6.0.20-summary.txt (added)
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@@ -1,0 +1,524 @@
+ Release Summary
+
+ asterisk-1.6.0.20
+
+ Date: 2009-12-17
+
+ <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-1.6.0.18.
+
+ ----------------------------------------------------------------------
+
+ 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
+ 13 tilghman 5 dvossel 3 falves11
+ 10 dvossel 5 falves11 2 alecdavis
+ 7 russell 3 alecdavis 2 atis
+ 4 kpfleming 3 mnicholson 2 dimas
+ 4 mnicholson 3 parisioa 2 parisioa
+ 3 alecdavis 3 tilghman 2 wdoekes
+ 2 atis 2 mnick 1 abelbeck
+ 2 dimas 1 abelbeck 1 ajohnson
+ 2 file 1 ajohnson 1 AlexMS
+ 2 jpeeler 1 AlexMS 1 amorsen
+ 2 mnick 1 amorsen 1 bklang
+ 2 oej 1 diLLec 1 CGMChris
+ 1 bklang 1 Laureano 1 corruptor
+ 1 diruggles 1 lmadsen 1 dant
+ 1 jsmith 1 oej 1 davidw
+ 1 Laureano 1 OrNix 1 dcabot
+ 1 lmadsen 1 Parantido 1 diLLec
+ 1 qwell 1 qwell 1 jiddings
+ 1 vnovy 1 rmudgett 1 mgernoth
+ 1 russell 1 msetim
+ 1 thedavidfactor 1 nicchap
+ 1 zalex1953 1 OrNix
+ 1 Parantido
+ 1 patrol-cz
+ 1 rmudgett
+ 1 sgimeno
+ 1 Shagg63
+ 1 udosw
+ 1 vnovy
+ 1 zalex1953
+
+ ----------------------------------------------------------------------
+
+ 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_amd
+
+ #16239: AMD() incorrectly sets AMDCAUSE channel variable
+ Revision: 232357
+ Reporter: CGMChris
+ Coders: file
+
+ Category: Applications/app_dial
+
+ #16337: [patch] Segmentation Fault on Originate command.
+ Revision: 232272
+ Reporter: Parantido
+ Testers: Parantido, dvossel
+ Coders: dvossel
+
+ Category: Applications/app_directory
+
+ #16437: [patch] Unable to escape back to dialplan or operator, using 'o'
+ and 'a' extensions in dialcontext
+ Revision: 234894
+ Reporter: alecdavis
+ Testers: alecdavis
+ Coders: alecdavis
+
+ Category: Applications/app_externalivr
+
+ #16305: [patch] ExternalIVR confuses AGI by double closing FDs
+ Revision: 232811
+ Reporter: diLLec
+ Testers: thedavidfactor, diLLec
+ Coders: diruggles
+
+ Category: Applications/app_meetme
+
+ #16247: [patch] Muted user remains talking forever
+ Revision: 234404
+ Reporter: dimas
+ Coders: dimas
+
+ Category: Applications/app_mixmonitor
+
+ #16152: [patch] MixMonitor thread doesn't exit until channel is dropped
+ Revision: 230512
+ Reporter: AlexMS
+ Testers: dvossel, AlexMS
+ Coders: dvossel
+
+ Category: Applications/app_voicemail
+
+ #15625: [patch] Prepending to a voicemail on forward causes locked sip
+ channel and large file filling disk space
+ Revision: 231691
+ Reporter: Shagg63
+ Testers: mnicholson
+ Coders: mnicholson
+
+ #16291: app_voicemail.c strip_control() strips more than just control
+ chars
+ Revision: 233167
+ Reporter: wdoekes
+ Coders: dvossel
+
+ Category: Channels/General
+
+ #16058: [patch] Crash in local_ast_moh_start / ast_indicate_data due to
+ AST_CONTROL_HOLD with bad pointer
+ Revision: 231928
+ Reporter: atis
+ Coders: jpeeler
+
+ #16242: [patch] Comfort noise frame with f->data NULL but f->datalen 160
+ Revision: 231517
+ Reporter: amorsen
+ Testers: amorsen, oej, dvossel
+ Coders: oej
+
+ Category: Channels/chan_agent
+
+ #14590: [patch] updatecdr='yes' in agents.conf is not working
+ Revision: 230631
+ Reporter: msetim
+ Testers: Laureano, mnicholson
+ Coders: Laureano
+
+ Category: Channels/chan_dahdi
+
+ #15972: [patch] Dropping frame since I'm still dialing on Zap/... (resp.
+ DAHDi/...) with DIGITAL calls
+ Revision: 232092
+ Reporter: udosw
+ Testers: alecdavis
+ Coders: alecdavis
+
+ Category: Channels/chan_iax2
+
+ #16094: [patch] iax2 show cache, locks channels.
+ Revision: 230729
+ Reporter: alecdavis
+ Testers: alecdavis, dvossel
+ Coders: alecdavis
+
+ Category: Channels/chan_sip/General
+
+ #15356: After a few thousand calls, or at random, Asterisk stops receiving
+ events from the network
+ Revision: 234131
+ Reporter: falves11
+ Testers: falves11
+ Coders: tilghman
+
+ #15716: [patch] chan_sip fails to destroy channels in INVITE when no
+ response received
+ Revision: 234131
+ Reporter: dant
+ Testers: falves11
+ Coders: tilghman
+
+ #16268: [patch] Last line of SDP is not being parsed
+ Revision: 230782
+ Reporter: sgimeno
+ Coders: kpfleming
+
+ Category: Channels/chan_sip/Interoperability
+
+ #16186: [patch] T.38 reinvite fails after receiving "415 Unsupported media
+ type" when it could continue in audio mode
+ Revision: 232346
+ Reporter: atis
+ Coders: atis
+
+ Category: Channels/chan_sip/Registration
+
+ #16298: [patch] After upgrading to asterisk 1.4.27 Optipoint SIP phone can
+ no longer register
+ Revision: 233475
+ Reporter: mgernoth
+ Testers: dvossel
+ Coders: dvossel
+
+ Category: Channels/chan_sip/T.38
+
+ #16387: [patch] Missing session level connection data (c=) breaks
+ process_sdp()
+ Revision: 233397
+ Reporter: zalex1953
+ Testers: mnicholson, zalex1953
+ Coders: mnicholson
+
+ Category: Core/BuildSystem
+
+ #14737: [patch] Cannot flavour (flavor) version number because
+ make_version_c looks in wrong directory
+ Revision: 234701
+ Reporter: davidw
+ Coders: tilghman
+
+ #16296: [patch] menuselect.makeopts: does not properly unselect an option
+ with a leading - (minus)
+ Revision: 234257
+ Reporter: abelbeck
+ Testers: abelbeck, qwell, lmadsen
+ Coders: qwell
+
+ Category: Core/General
+
+ #15769: [patch] useless message pops hundreds of times per minute
+ Revision: 233047
+ Reporter: falves11
+ Testers: mnick, falves11
+ Coders: mnick
+
+ #16106: [patch] Hangup extension executed twice in 1.6.2 RC2
+ Revision: 231096
+ Reporter: ajohnson
+ Testers: ajohnson
+ Coders: jpeeler
+
+ #16290: ast_ouraddrfor doesn't do htons() on the port
+ Revision: 232354
+ Reporter: wdoekes
+ Coders: dvossel
+
+ Category: Core/ManagerInterface
+
+ #16264: [patch] UserEvent manager action is not ACKed
+ Revision: 232583
+ Reporter: dimas
+ Coders: dimas
+
+ #16275: [patch] response to "Action: Events" is not finished by empty line
+ Revision: 232577
+ Reporter: vnovy
+ Coders: vnovy
+
+ Category: Core/Netsock
+
+ #15627: [patch] Asterisk runs out of sockets
+ Revision: 234131
+ Reporter: falves11
+ Testers: falves11
+ Coders: tilghman
+
+ #16270: [patch] Asterisk doesn't free udp ports
+ Revision: 234131
+ Reporter: corruptor
+ Testers: falves11
+ Coders: tilghman
+
+ Category: Documentation
+
+ #16223: [patch] "requirecalltoken" config directive not respected globally
+ Revision: 233284
+ Reporter: bklang
+ Coders: bklang
+
+ Category: Formats/General
+
+ #16412: Ignoring unknown format wav & wav49...
+ Revision: 233841
+ Reporter: jiddings
+ Testers: russell
+ Coders: russell
+
+ Category: General
+
+ #16113: [patch] Auto-fallthrough when attempting to enter DTMF using
+ Background() in a Macro()
+ Revision: 235423
+ Reporter: OrNix
+ Testers: OrNix
+ Coders: tilghman
+
+ #16272: [patch] Language code collisions for certan languages
+ Revision: 232864
+ Reporter: patrol-cz
+ Coders: tilghman
+
+ #16446: segfault error 4
+ Revision: 235054
+ Reporter: nicchap
+ Coders: tilghman
+
+ Category: PBX/General
+
+ #16166: [patch] Setting dialplan hint and using a global variable gives
+ incorrect warning.
+ Revision: 233236
+ Reporter: rmudgett
+ Testers: mnick, rmudgett
+ Coders: mnick
+
+ Category: Resources/res_musiconhold
+
+ #16207: [patch] asterisk keeps starting new processes for streaming audio
+ MOH
+ Revision: 232699
+ Reporter: dcabot
+ Testers: parisioa, tilghman
+ Coders: tilghman
+
+ #16279: [patch] asterisk reload causes mpg123 streams to be recreated
+ Revision: 232699
+ Reporter: parisioa
+ Testers: parisioa, tilghman
+ Coders: tilghman
+
+ #16388: [patch] New music on hold patches cause asterisk + full system
+ hard lock
+ Revision: 233729
+ Reporter: parisioa
+ Testers: parisioa, tilghman
+ Coders: tilghman
+
+ ----------------------------------------------------------------------
+
+ 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 |
+ |----------+------------+-----------------------------------+------------|
+ | 230587 | dvossel | audiohook signal trigger on every | #14618 |
+ | | | status change | |
+ |----------+------------+-----------------------------------+------------|
+ | | | Correct fix for issue #16268... | |
+ | 230878 | kpfleming | the reporter's original patch was | #16268 |
[... 153 lines stripped ...]
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