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Asterisk Autobuilder
7f4361bf9c Importing release summary for 1.8.15.0 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0@370561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30 16:43:25 +00:00
Asterisk Autobuilder
bc448fab84 Update version, ChangeLog, remove old summaries
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0@370559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30 16:40:58 +00:00
Asterisk Autobuilder
74f504ee1e Create 1.8.15.0
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0@370556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30 16:38:57 +00:00
Asterisk Autobuilder
3b69ca9d9c Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0-rc1@369925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 15:42:33 +00:00
Asterisk Autobuilder
1e8c2db8f9 Importing release summary for 1.8.15.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0-rc1@369924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 15:42:25 +00:00
Asterisk Autobuilder
60c89f482e Importing files for 1.8.15.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0-rc1@369923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 15:42:18 +00:00
Asterisk Autobuilder
9622e36d23 Creating tag for the release of asterisk-1.8.15.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0-rc1@369922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 15:40:15 +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-1.8.15.0</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-1.8.15.0</h3>
<h3 align="center">Date: 2012-07-30</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-1.8.14.0.</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>
8 rmudgett<br/>
7 mjordan<br/>
7 mmichelson<br/>
5 kmoore<br/>
4 Mark<br/>
4 twilson<br/>
3 may<br/>
2 jrose<br/>
2 kpfleming<br/>
1 file<br/>
1 jcolp<br/>
1 Michael<br/>
1 qwell<br/>
</td>
<td>
2 Steve Davies<br/>
2 Terry Wilson<br/>
1 Dan Delaney<br/>
1 Guenther Kelleter<br/>
1 jamicque<br/>
1 Julian Yap<br/>
1 Michael L. Young<br/>
1 Paul Belanger<br/>
1 rmudgett<br/>
1 Tilghman Lesher<br/>
</td>
<td>
3 lmadsen<br/>
2 fnordian<br/>
2 one47<br/>
1 alecdavis<br/>
1 drdelaney<br/>
1 elguero<br/>
1 jamicque<br/>
1 karlfife<br/>
1 mdavenport<br/>
1 mjordan<br/>
1 mmichelson<br/>
1 sdolloff<br/>
1 themsley<br/>
1 tomaso<br/>
1 tsarik<br/>
1 twilson<br/>
1 vsauer<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: Addons/chan_ooh323</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369090">369090</a><br/>
Reporter: tsarik<br/>
Coders: may<br/>
<br/>
<h3>Category: Applications/app_dial</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19611">ASTERISK-19611</a>: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368898">368898</a><br/>
Reporter: vsauer<br/>
Coders: Mark<br/>
<br/>
<h3>Category: Applications/app_voicemail</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19923">ASTERISK-19923</a>: Asterisk crashing due to memory corruptions in chan_sip/voicemail<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369652">369652</a><br/>
Reporter: drdelaney<br/>
Testers: Dan Delaney, Julian Yap<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: Channels/chan_iax2</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19801">ASTERISK-19801</a>: Deadlock with masquerade and chan_iax<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368759">368759</a><br/>
Reporter: alecdavis<br/>
Testers: Guenther Kelleter<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_local</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19611">ASTERISK-19611</a>: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368898">368898</a><br/>
Reporter: vsauer<br/>
Coders: Mark<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369090">369090</a><br/>
Reporter: tsarik<br/>
Coders: may<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19859">ASTERISK-19859</a>: cid_tag is not set according to the sip configuration anymore if get_rpid() != 0<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368807">368807</a><br/>
Reporter: tomaso<br/>
Coders: mmichelson<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19892">ASTERISK-19892</a>: If Asterisk sends a 481 to an initial INVITE that contained a to-tag, then Asterisk will not recognize the ensuing ACK<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369352">369352</a><br/>
Reporter: mmichelson<br/>
Coders: mmichelson<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369436">369436</a><br/>
Reporter: one47<br/>
Testers: Steve Davies, Terry Wilson<br/>
Coders: twilson<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369557">369557</a><br/>
Reporter: one47<br/>
Testers: Steve Davies, Terry Wilson<br/>
Coders: twilson<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20008">ASTERISK-20008</a>: outboundproxy ignored after when sending invite after 407<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369066">369066</a><br/>
Reporter: fnordian<br/>
Coders: Mark<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20040">ASTERISK-20040</a>: Asterisk crashes when a guest call uses directmedia<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369214">369214</a><br/>
Reporter: twilson<br/>
Coders: twilson<br/>
<br/>
<h3>Category: Channels/chan_sip/IPv6</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16618">ASTERISK-16618</a>: Unable to use IPv4 addresses for a TCP host when using IPv6<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369471">369471</a><br/>
Reporter: lmadsen<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19601">ASTERISK-19601</a>: Failure of domain matching on authentication of INVITE request produces misleading NOTICE message; bypasses alwaysauthreject logic<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369302">369302</a><br/>
Reporter: mjordan<br/>
Coders: Mark<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19910">ASTERISK-19910</a>: Add sip_notify.conf entry for Digium phones<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369818">369818</a><br/>
Reporter: mdavenport<br/>
Coders: qwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19920">ASTERISK-19920</a>: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368873">368873</a><br/>
Reporter: lmadsen<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Core/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19834">ASTERISK-19834</a>: Memory leak caused by thread created by bridge_channel_join being neither joined nor detached<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369708">369708</a><br/>
Reporter: fnordian<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Core/Netsock</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20006">ASTERISK-20006</a>: Fix NULL pointer segfault in ast_sockaddr_parse()<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369108">369108</a><br/>
Reporter: elguero<br/>
Testers: Michael L. Young<br/>
Coders: Michael<br/>
<br/>
<h3>Category: Documentation</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20007">ASTERISK-20007</a>: GotoIf() documentation updates to be more clear that [[context,]extension,]priority is valid<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369869">369869</a><br/>
Reporter: lmadsen<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19492">ASTERISK-19492</a>: Group write permission removed from existing directory /etc/asterisk/. when updating <br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368830">368830</a><br/>
Reporter: karlfife<br/>
Testers: Paul Belanger, Tilghman Lesher<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_adsi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19920">ASTERISK-19920</a>: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368873">368873</a><br/>
Reporter: lmadsen<br/>
Coders: mmichelson<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/1.8?view=revision&revision=368719">368719</a></td><td>kmoore</td><td>Fix compilation in dev-mode</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368738">368738</a></td><td>kmoore</td><td>Fix coverity UNUSED_VALUE findings in core support level files</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19672">ASTERISK-19672</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368852">368852</a></td><td>mjordan</td><td>Do not install empty directories; add ASTLIBDIR</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368894">368894</a></td><td>mjordan</td><td>Mark res_smdi/res_adsi as 'core' supported modules</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368927">368927</a></td><td>mmichelson</td><td>Revert Makefile change to remove embedding res_adsi.so</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369001">369001</a></td><td>kpfleming</td><td>Add support-level indications to many more source files.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369002">369002</a></td><td>kpfleming</td><td>Add a script to enable finding source files without support-levels defined.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369130">369130</a></td><td>may</td><td>fix compile error (1.8 don't have ast_channel_name macro)</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369146">369146</a></td><td>may</td><td>fix locking issue on empty callList</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19298">ASTERISK-19298</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369195">369195</a></td><td>kmoore</td><td>Don't parse media stream state for SIP video streams</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369235">369235</a></td><td>rmudgett</td><td>Change incorrect chan_sip zombie hangup debug message. They are all zombies now.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369238">369238</a></td><td>rmudgett</td><td>Check if PBX was started for generic CCSS recall.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369258">369258</a></td><td>rmudgett</td><td>Check if PBX was started and fix F and F(x) action logic in Dial application.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369262">369262</a></td><td>rmudgett</td><td>Explicitly check caller hangup in app Queue rather than a polluted res2 value.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369282">369282</a></td><td>rmudgett</td><td>Fix Bridge application and AMI Bridge action error handling.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369323">369323</a></td><td>mmichelson</td><td>Eliminate embedding of res_adsi.so module.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369324">369324</a></td><td>mmichelson</td><td>Forgot to svn add this file in my last commit.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369351">369351</a></td><td>mjordan</td><td>Fix incorrect duration reporting in CDRs created in batch mode</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19860">ASTERISK-19860</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369366">369366</a></td><td>mjordan</td><td>Tweak CDR change in r369351</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369390">369390</a></td><td>mjordan</td><td>Fix crash in unloading of res_adsi module</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369490">369490</a></td><td>file</td><td>With some configurations a transport is not actually specified so assume UDP in these cases.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369579">369579</a></td><td>twilson</td><td>More improvements to re-INVITEs timing out after a provisional response</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369626">369626</a></td><td>mjordan</td><td>Do not send a BYE when a provisional response arrives during a re-INVITE</td>
<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369750">369750</a></td><td>jrose</td><td>chan_sip: Add case for FLASH control frames so that we don't display a warning.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369792">369792</a></td><td>jrose</td><td>chan_sip: Fix small behavioral change accidentally introduced in r369750</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>
Makefile | 46 +--
addons/chan_ooh323.c | 23 +
addons/ooh323c/src/ooCalls.c | 3
addons/ooh323c/src/ooq931.c | 2
apps/app_dial.c | 34 +-
apps/app_directory.c | 3
apps/app_queue.c | 14 -
apps/app_stack.c | 5
apps/app_voicemail.c | 85 +++++-
build_tools/find_missing_support_level | 3
channels/chan_dahdi.c | 16 -
channels/chan_iax2.c | 15 -
channels/chan_misdn.c | 1
channels/chan_sip.c | 233 +++++++++++++-----
channels/console_board.c | 4
channels/console_gui.c | 4
channels/console_video.c | 4
channels/iax2-parser.c | 4
channels/iax2-provision.c | 4
channels/misdn/ie.c | 4
channels/misdn/isdn_lib.c | 4
channels/misdn/isdn_msg_parser.c | 4
channels/misdn/portinfo.c | 3
channels/misdn_config.c | 4
channels/sig_analog.c | 15 +
channels/sig_pri.c | 3
channels/sig_ss7.c | 3
channels/sip/config_parser.c | 4
channels/sip/dialplan_functions.c | 8
channels/sip/include/sip.h | 4
channels/sip/reqresp_parser.c | 6
channels/sip/sdp_crypto.c | 8
channels/sip/srtp.c | 4
channels/vcodecs.c | 4
channels/vgrabbers.c | 4
configs/sip_notify.conf.sample | 5
funcs/func_strings.c | 3
funcs/func_volume.c | 3
include/asterisk/adsi.h | 93 +++++--
include/asterisk/channel.h | 2
include/asterisk/netsock2.h | 3
main/Makefile | 3
main/abstract_jb.c | 4
main/acl.c | 4
main/adsi.c | 351 ++++++++++++++++++++++++++++
main/alaw.c | 4
main/aoc.c | 4
main/app.c | 4
main/asterisk.c | 4
main/astfd.c | 4
main/astmm.c | 4
main/astobj2.c | 5
main/audiohook.c | 4
main/autochan.c | 4
main/autoservice.c | 4
main/bridging.c | 18 -
main/callerid.c | 4
main/ccss.c | 13 -
main/cdr.c | 10
main/cel.c | 4
main/channel.c | 14 -
main/chanvars.c | 4
main/cli.c | 4
main/config.c | 4
main/data.c | 4
main/datastore.c | 4
main/db.c | 4
main/devicestate.c | 4
main/dial.c | 4
main/dns.c | 4
main/dnsmgr.c | 4
main/dsp.c | 4
main/enum.c | 4
main/event.c | 4
main/features.c | 409 ++++++++++++++++++---------------
main/file.c | 4
main/fixedjitterbuf.c | 4
main/frame.c | 4
main/framehook.c | 4
main/fskmodem.c | 4
main/fskmodem_float.c | 4
main/fskmodem_int.c | 4
main/global_datastores.c | 4
main/hashtab.c | 4
main/heap.c | 4
main/image.c | 4
main/indications.c | 4
main/io.c | 4
main/jitterbuf.c | 4
main/loader.c | 8
main/lock.c | 4
main/logger.c | 4
main/md5.c | 6
main/netsock.c | 4
main/netsock2.c | 10
main/pbx.c | 24 +
main/plc.c | 4
main/privacy.c | 4
main/rtp_engine.c | 4
main/say.c | 6
main/sched.c | 4
main/security_events.c | 4
main/slinfactory.c | 4
main/srv.c | 4
main/ssl.c | 4
main/stdtime/localtime.c | 4
main/strcompat.c | 4
main/strings.c | 4
main/stun.c | 4
main/syslog.c | 4
main/taskprocessor.c | 4
main/tcptls.c | 7
main/tdd.c | 4
main/term.c | 4
main/test.c | 4
main/threadstorage.c | 4
main/timing.c | 4
main/translate.c | 4
main/udptl.c | 7
main/ulaw.c | 4
main/utils.c | 4
main/xml.c | 4
main/xmldoc.c | 4
pbx/dundi-parser.c | 4
pbx/pbx_config.c | 4
res/ael/pval.c | 4
res/ais/clm.c | 4
res/ais/evt.c | 4
res/res_adsi.c | 187 ++++++++++-----
res/res_adsi.exports.in | 33 --
res/res_config_odbc.c | 7
res/res_fax.c | 2
res/res_odbc.c | 2
res/res_smdi.c | 2
res/res_speech.c | 3
res/snmp/agent.c | 4
136 files changed, 1597 insertions(+), 516 deletions(-)
</pre><br/>
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Release Summary
asterisk-1.8.15.0
Date: 2012-07-30
<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-1.8.14.0.
----------------------------------------------------------------------
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
8 rmudgett 2 Steve Davies 3 lmadsen
7 mjordan 2 Terry Wilson 2 fnordian
7 mmichelson 1 Dan Delaney 2 one47
5 kmoore 1 Guenther Kelleter 1 alecdavis
4 Mark 1 jamicque 1 drdelaney
4 twilson 1 Julian Yap 1 elguero
3 may 1 Michael L. Young 1 jamicque
2 jrose 1 Paul Belanger 1 karlfife
2 kpfleming 1 rmudgett 1 mdavenport
1 file 1 Tilghman Lesher 1 mjordan
1 jcolp 1 mmichelson
1 Michael 1 sdolloff
1 qwell 1 themsley
1 tomaso
1 tsarik
1 twilson
1 vsauer
----------------------------------------------------------------------
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: Addons/chan_ooh323
ASTERISK-19179: RTP inactivity SIP / ooh323 wont work
Revision: 369090
Reporter: tsarik
Coders: may
Category: Applications/app_dial
ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom
phone is redirected by "302 Moved temporarily" to chan_local
Revision: 368898
Reporter: vsauer
Coders: Mark
Category: Applications/app_voicemail
ASTERISK-19923: Asterisk crashing due to memory corruptions in
chan_sip/voicemail
Revision: 369652
Reporter: drdelaney
Testers: Dan Delaney, Julian Yap
Coders: kmoore
Category: Channels/chan_iax2
ASTERISK-19801: Deadlock with masquerade and chan_iax
Revision: 368759
Reporter: alecdavis
Testers: Guenther Kelleter
Coders: rmudgett
Category: Channels/chan_local
ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom
phone is redirected by "302 Moved temporarily" to chan_local
Revision: 368898
Reporter: vsauer
Coders: Mark
Category: Channels/chan_sip/General
ASTERISK-19179: RTP inactivity SIP / ooh323 wont work
Revision: 369090
Reporter: tsarik
Coders: may
ASTERISK-19859: cid_tag is not set according to the sip configuration
anymore if get_rpid() != 0
Revision: 368807
Reporter: tomaso
Coders: mmichelson
ASTERISK-19892: If Asterisk sends a 481 to an initial INVITE that
contained a to-tag, then Asterisk will not recognize the ensuing ACK
Revision: 369352
Reporter: mmichelson
Coders: mmichelson
ASTERISK-19992: SIP re-INVITEs have no transaction timeout
Revision: 369436
Reporter: one47
Testers: Steve Davies, Terry Wilson
Coders: twilson
ASTERISK-19992: SIP re-INVITEs have no transaction timeout
Revision: 369557
Reporter: one47
Testers: Steve Davies, Terry Wilson
Coders: twilson
ASTERISK-20008: outboundproxy ignored after when sending invite after 407
Revision: 369066
Reporter: fnordian
Coders: Mark
ASTERISK-20040: Asterisk crashes when a guest call uses directmedia
Revision: 369214
Reporter: twilson
Coders: twilson
Category: Channels/chan_sip/IPv6
ASTERISK-16618: Unable to use IPv4 addresses for a TCP host when using
IPv6
Revision: 369471
Reporter: lmadsen
Coders: jcolp
Category: Channels/chan_sip/Interoperability
ASTERISK-19601: Failure of domain matching on authentication of INVITE
request produces misleading NOTICE message; bypasses alwaysauthreject
logic
Revision: 369302
Reporter: mjordan
Coders: Mark
Category: Core/Configuration
ASTERISK-19910: Add sip_notify.conf entry for Digium phones
Revision: 369818
Reporter: mdavenport
Coders: qwell
ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is)
despite no modules.conf, noload or autoload=no instructions
Revision: 368873
Reporter: lmadsen
Coders: mmichelson
Category: Core/General
ASTERISK-19834: Memory leak caused by thread created by
bridge_channel_join being neither joined nor detached
Revision: 369708
Reporter: fnordian
Coders: mmichelson
Category: Core/Netsock
ASTERISK-20006: Fix NULL pointer segfault in ast_sockaddr_parse()
Revision: 369108
Reporter: elguero
Testers: Michael L. Young
Coders: Michael
Category: Documentation
ASTERISK-20007: GotoIf() documentation updates to be more clear that
[[context,]extension,]priority is valid
Revision: 369869
Reporter: lmadsen
Coders: kmoore
Category: General
ASTERISK-19492: Group write permission removed from existing directory
/etc/asterisk/. when updating
Revision: 368830
Reporter: karlfife
Testers: Paul Belanger, Tilghman Lesher
Coders: mjordan
Category: Resources/res_adsi
ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is)
despite no modules.conf, noload or autoload=no instructions
Revision: 368873
Reporter: lmadsen
Coders: mmichelson
----------------------------------------------------------------------
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 |
|----------+------------+----------------------------+-------------------|
| 368719 | kmoore | Fix compilation in | |
| | | dev-mode | |
|----------+------------+----------------------------+-------------------|
| | | Fix coverity UNUSED_VALUE | |
| 368738 | kmoore | findings in core support | ASTERISK-19672 |
| | | level files | |
|----------+------------+----------------------------+-------------------|
| 368852 | mjordan | Do not install empty | |
| | | directories; add ASTLIBDIR | |
|----------+------------+----------------------------+-------------------|
| 368894 | mjordan | Mark res_smdi/res_adsi as | |
| | | 'core' supported modules | |
|----------+------------+----------------------------+-------------------|
| | | Revert Makefile change to | |
| 368927 | mmichelson | remove embedding | |
| | | res_adsi.so | |
|----------+------------+----------------------------+-------------------|
| | | Add support-level | |
| 369001 | kpfleming | indications to many more | |
| | | source files. | |
|----------+------------+----------------------------+-------------------|
| | | Add a script to enable | |
| 369002 | kpfleming | finding source files | |
| | | without support-levels | |
| | | defined. | |
|----------+------------+----------------------------+-------------------|
| | | fix compile error (1.8 | |
| 369130 | may | don't have | |
| | | ast_channel_name macro) | |
|----------+------------+----------------------------+-------------------|
| 369146 | may | fix locking issue on empty | ASTERISK-19298 |
| | | callList | |
|----------+------------+----------------------------+-------------------|
| | | Don't parse media stream | |
| 369195 | kmoore | state for SIP video | |
| | | streams | |
|----------+------------+----------------------------+-------------------|
| | | Change incorrect chan_sip | |
| 369235 | rmudgett | zombie hangup debug | |
| | | message. They are all | |
| | | zombies now. | |
|----------+------------+----------------------------+-------------------|
| 369238 | rmudgett | Check if PBX was started | |
| | | for generic CCSS recall. | |
|----------+------------+----------------------------+-------------------|
| | | Check if PBX was started | |
| 369258 | rmudgett | and fix F and F(x) action | |
| | | logic in Dial application. | |
|----------+------------+----------------------------+-------------------|
| | | Explicitly check caller | |
| 369262 | rmudgett | hangup in app Queue rather | |
| | | than a polluted res2 | |
| | | value. | |
|----------+------------+----------------------------+-------------------|
| | | Fix Bridge application and | |
| 369282 | rmudgett | AMI Bridge action error | |
| | | handling. | |
|----------+------------+----------------------------+-------------------|
| 369323 | mmichelson | Eliminate embedding of | |
| | | res_adsi.so module. | |
|----------+------------+----------------------------+-------------------|
| 369324 | mmichelson | Forgot to svn add this | |
| | | file in my last commit. | |
|----------+------------+----------------------------+-------------------|
| | | Fix incorrect duration | |
| 369351 | mjordan | reporting in CDRs created | ASTERISK-19860 |
| | | in batch mode | |
|----------+------------+----------------------------+-------------------|
| 369366 | mjordan | Tweak CDR change in | |
| | | r369351 | |
|----------+------------+----------------------------+-------------------|
| 369390 | mjordan | Fix crash in unloading of | |
| | | res_adsi module | |
|----------+------------+----------------------------+-------------------|
| | | With some configurations a | |
| 369490 | file | transport is not actually | |
| | | specified so assume UDP in | |
| | | these cases. | |
|----------+------------+----------------------------+-------------------|
| | | More improvements to | |
| 369579 | twilson | re-INVITEs timing out | ASTERISK-19992 |
| | | after a provisional | |
| | | response | |
|----------+------------+----------------------------+-------------------|
| | | Do not send a BYE when a | |
| 369626 | mjordan | provisional response | ASTERISK-19992 |
| | | arrives during a re-INVITE | |
|----------+------------+----------------------------+-------------------|
| | | chan_sip: Add case for | |
| 369750 | jrose | FLASH control frames so | |
| | | that we don't display a | |
| | | warning. | |
|----------+------------+----------------------------+-------------------|
| | | chan_sip: Fix small | |
| 369792 | jrose | behavioral change | |
| | | accidentally introduced in | |
| | | r369750 | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
Makefile | 46 +--
addons/chan_ooh323.c | 23 +
addons/ooh323c/src/ooCalls.c | 3
addons/ooh323c/src/ooq931.c | 2
apps/app_dial.c | 34 +-
apps/app_directory.c | 3
apps/app_queue.c | 14 -
apps/app_stack.c | 5
apps/app_voicemail.c | 85 +++++-
build_tools/find_missing_support_level | 3
channels/chan_dahdi.c | 16 -
channels/chan_iax2.c | 15 -
channels/chan_misdn.c | 1
channels/chan_sip.c | 233 +++++++++++++-----
channels/console_board.c | 4
channels/console_gui.c | 4
channels/console_video.c | 4
channels/iax2-parser.c | 4
channels/iax2-provision.c | 4
channels/misdn/ie.c | 4
channels/misdn/isdn_lib.c | 4
channels/misdn/isdn_msg_parser.c | 4
channels/misdn/portinfo.c | 3
channels/misdn_config.c | 4
channels/sig_analog.c | 15 +
channels/sig_pri.c | 3
channels/sig_ss7.c | 3
channels/sip/config_parser.c | 4
channels/sip/dialplan_functions.c | 8
channels/sip/include/sip.h | 4
channels/sip/reqresp_parser.c | 6
channels/sip/sdp_crypto.c | 8
channels/sip/srtp.c | 4
channels/vcodecs.c | 4
channels/vgrabbers.c | 4
configs/sip_notify.conf.sample | 5
funcs/func_strings.c | 3
funcs/func_volume.c | 3
include/asterisk/adsi.h | 93 +++++--
include/asterisk/channel.h | 2
include/asterisk/netsock2.h | 3
main/Makefile | 3
main/abstract_jb.c | 4
main/acl.c | 4
main/adsi.c | 351 ++++++++++++++++++++++++++++
main/alaw.c | 4
main/aoc.c | 4
main/app.c | 4
main/asterisk.c | 4
main/astfd.c | 4
main/astmm.c | 4
main/astobj2.c | 5
main/audiohook.c | 4
main/autochan.c | 4
main/autoservice.c | 4
main/bridging.c | 18 -
main/callerid.c | 4
main/ccss.c | 13 -
main/cdr.c | 10
main/cel.c | 4
main/channel.c | 14 -
main/chanvars.c | 4
main/cli.c | 4
main/config.c | 4
main/data.c | 4
main/datastore.c | 4
main/db.c | 4
main/devicestate.c | 4
main/dial.c | 4
main/dns.c | 4
main/dnsmgr.c | 4
main/dsp.c | 4
main/enum.c | 4
main/event.c | 4
main/features.c | 409 ++++++++++++++++++---------------
main/file.c | 4
main/fixedjitterbuf.c | 4
main/frame.c | 4
main/framehook.c | 4
main/fskmodem.c | 4
main/fskmodem_float.c | 4
main/fskmodem_int.c | 4
main/global_datastores.c | 4
main/hashtab.c | 4
main/heap.c | 4
main/image.c | 4
main/indications.c | 4
main/io.c | 4
main/jitterbuf.c | 4
main/loader.c | 8
main/lock.c | 4
main/logger.c | 4
main/md5.c | 6
main/netsock.c | 4
main/netsock2.c | 10
main/pbx.c | 24 +
main/plc.c | 4
main/privacy.c | 4
main/rtp_engine.c | 4
main/say.c | 6
main/sched.c | 4
main/security_events.c | 4
main/slinfactory.c | 4
main/srv.c | 4
main/ssl.c | 4
main/stdtime/localtime.c | 4
main/strcompat.c | 4
main/strings.c | 4
main/stun.c | 4
main/syslog.c | 4
main/taskprocessor.c | 4
main/tcptls.c | 7
main/tdd.c | 4
main/term.c | 4
main/test.c | 4
main/threadstorage.c | 4
main/timing.c | 4
main/translate.c | 4
main/udptl.c | 7
main/ulaw.c | 4
main/utils.c | 4
main/xml.c | 4
main/xmldoc.c | 4
pbx/dundi-parser.c | 4
pbx/pbx_config.c | 4
res/ael/pval.c | 4
res/ais/clm.c | 4
res/ais/evt.c | 4
res/res_adsi.c | 187 ++++++++++-----
res/res_adsi.exports.in | 33 --
res/res_config_odbc.c | 7
res/res_fax.c | 2
res/res_odbc.c | 2
res/res_smdi.c | 2
res/res_speech.c | 3
res/snmp/agent.c | 4
136 files changed, 1597 insertions(+), 516 deletions(-)
----------------------------------------------------------------------