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Asterisk Autobuilder
e14cc5edb2 Importing release summary for 11.1.0 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0@377508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-10 01:33:00 +00:00
Asterisk Autobuilder
bec79e9cee Update version, ChangeLog
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0@377484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-10 01:00:39 +00:00
Asterisk Autobuilder
72be8fdbc7 Create 11.1.0
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0@377480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-10 00:35:27 +00:00
Asterisk Autobuilder
6e27c017c7 Importing release summary for 11.1.0-rc3 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc3@377328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-06 14:33:23 +00:00
Asterisk Autobuilder
f28acbe753 Merge r376870 for 11.1.0-rc3
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc3@377323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-06 14:25:59 +00:00
Asterisk Autobuilder
93c77321ca Create 11.1.0-rc3
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc3@377318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-06 13:55:39 +00:00
Asterisk Autobuilder
249bb3e729 Importing release summary for 11.1.0-rc2 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc2@377302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 18:05:43 +00:00
Asterisk Autobuilder
4fded6e1b6 Merge r377259 for 11.1.0-rc2
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc2@377299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 17:58:06 +00:00
Asterisk Autobuilder
b9b6bd9b5b Create 11.1.0-rc2
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc2@377293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 17:34:55 +00:00
Asterisk Autobuilder
c8107eefe9 Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc1@375963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 16:34:31 +00:00
Asterisk Autobuilder
34434b6edc Importing release summary for 11.1.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc1@375962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 16:34:21 +00:00
Asterisk Autobuilder
262df6b3d5 Importing files for 11.1.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc1@375961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 16:34:14 +00:00
Asterisk Autobuilder
c88e3db5e3 Creating tag for the release of asterisk-11.1.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.1.0-rc1@375960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 16:32:12 +00:00
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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">&lt;asteriskteam@digium.com&gt;</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>

480
asterisk-11.1.0-summary.txt Normal file
View File

@@ -0,0 +1,480 @@
Release Summary
asterisk-11.1.0
Date: 2012-12-09
<asteriskteam@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 |
|----------+------------+----------------------------+-------------------|
| | | Prevent multiple CDR | |
| 375729 | mjordan | batches from conflicting | |
| | | when scheduling the CDR | |
| | | write | |
|----------+------------+----------------------------+-------------------|
| | | Only deref a reserved | |
| 375798 | mjordan | gateway session if we | |
| | | actually reserved one | |
|----------+------------+----------------------------+-------------------|
| | | Don't attempt to purge | |
| 375802 | mjordan | sessions when no sessions | |
| | | exist | |
|----------+------------+----------------------------+-------------------|
| | | Add safety NULL pointer | |
| 375864 | rmudgett | check in module user | |
| | | references. | |
|----------+------------+----------------------------+-------------------|
| | | Refactor ast_timer_ack to | |
| 375895 | mjordan | return an error and handle | ASTERISK-20032 |
| | | the error in timer users | |
|----------+------------+----------------------------+-------------------|
| | | Fix a bug where our Motif | |
| 375925 | file | ICE candidates were not | |
| | | quite proper, and make us | |
| | | more forgiving. | |
|----------+------------+----------------------------+-------------------|
| 377293 | bebuild | Create 11.1.0-rc2 | |
|----------+------------+----------------------------+-------------------|
| 377299 | bebuild | Merge r377259 for | |
| | | 11.1.0-rc2 | |
|----------+------------+----------------------------+-------------------|
| 377302 | bebuild | Importing release summary | |
| | | for 11.1.0-rc2 release. | |
|----------+------------+----------------------------+-------------------|
| 377318 | bebuild | Create 11.1.0-rc3 | |
|----------+------------+----------------------------+-------------------|
| 377323 | bebuild | Merge r376870 for | |
| | | 11.1.0-rc3 | |
|----------+------------+----------------------------+-------------------|
| 377328 | bebuild | Importing release summary | |
| | | for 11.1.0-rc3 release. | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
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(-)
----------------------------------------------------------------------

View File

@@ -311,15 +311,20 @@ static int local_devicestate(const char *data)
res = AST_DEVICE_NOT_INUSE;
it = ao2_iterator_init(locals, 0);
while ((lp = ao2_iterator_next(&it)) && (res == AST_DEVICE_NOT_INUSE)) {
if (!strcmp(exten, lp->exten) && !strcmp(context, lp->context) && lp->owner) {
ao2_lock(lp);
if (ast_test_flag(lp, LOCAL_LAUNCHED_PBX)) {
res = AST_DEVICE_INUSE;
}
ao2_unlock(lp);
for (; (lp = ao2_iterator_next(&it)); ao2_ref(lp, -1)) {
int is_inuse;
ao2_lock(lp);
is_inuse = !strcmp(exten, lp->exten)
&& !strcmp(context, lp->context)
&& lp->owner
&& ast_test_flag(lp, LOCAL_LAUNCHED_PBX);
ao2_unlock(lp);
if (is_inuse) {
res = AST_DEVICE_INUSE;
ao2_ref(lp, -1);
break;
}
ao2_ref(lp, -1);
}
ao2_iterator_destroy(&it);

View File

@@ -2663,10 +2663,10 @@ static int sip_check_authtimeout(time_t start)
* \retval -1 Failed to read data
* \retval 0 Succeeded in reading data
*/
static int sip_tls_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session, int authenticated, time_t start, struct sip_threadinfo *me)
static int sip_tls_read(struct sip_request *req, struct sip_request *reqcpy, struct ast_tcptls_session_instance *tcptls_session,
int authenticated, time_t start, struct sip_threadinfo *me)
{
int res, content_length, after_poll = 1, need_poll = 1;
struct sip_request reqcpy = { 0, };
char buf[1024] = "";
int timeout = -1;
@@ -2720,10 +2720,10 @@ static int sip_tls_read(struct sip_request *req, struct ast_tcptls_session_insta
}
ast_str_append(&req->data, 0, "%s", buf);
}
copy_request(&reqcpy, req);
parse_request(&reqcpy);
copy_request(reqcpy, req);
parse_request(reqcpy);
/* In order to know how much to read, we need the content-length header */
if (sscanf(sip_get_header(&reqcpy, "Content-Length"), "%30d", &content_length)) {
if (sscanf(sip_get_header(reqcpy, "Content-Length"), "%30d", &content_length)) {
while (content_length > 0) {
size_t bytes_read;
if (!tcptls_session->client && !authenticated) {
@@ -3136,7 +3136,7 @@ static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_s
}
req.socket.fd = tcptls_session->fd;
if (tcptls_session->ssl) {
res = sip_tls_read(&req, tcptls_session, authenticated, start, me);
res = sip_tls_read(&req, &reqcpy, tcptls_session, authenticated, start, me);
} else {
res = sip_tcp_read(&req, tcptls_session, authenticated, start);
}