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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.15.0-rc1@369924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
383 lines
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383 lines
24 KiB
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<html xmlns="http://www.w3.org/1999/xhtml">
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<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-1.8.15.0-rc1</title></head>
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<body>
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<h1 align="center"><a name="top">Release Summary</a></h1>
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<h3 align="center">asterisk-1.8.15.0-rc1</h3>
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<h3 align="center">Date: 2012-07-11</h3>
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<h3 align="center"><asteriskteam@digium.com></h3>
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<hr/>
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<h2 align="center">Table of Contents</h2>
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<ol>
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<li><a href="#summary">Summary</a></li>
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<li><a href="#contributors">Contributors</a></li>
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<li><a href="#issues">Closed Issues</a></li>
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<li><a href="#commits">Other Changes</a></li>
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<li><a href="#diffstat">Diffstat</a></li>
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</ol>
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<hr/>
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<a name="summary"><h2 align="center">Summary</h2></a>
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<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>
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<p>The data in this summary reflects changes that have been made since the previous release, asterisk-1.8.14.0.</p>
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<hr/>
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<a name="contributors"><h2 align="center">Contributors</h2></a>
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<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>
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<table width="100%" border="0">
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<tr>
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<td width="33%"><h3>Coders</h3></td>
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<td width="33%"><h3>Testers</h3></td>
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<td width="33%"><h3>Reporters</h3></td>
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</tr>
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<tr valign="top">
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<td>
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8 rmudgett<br/>
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7 mjordan<br/>
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7 mmichelson<br/>
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5 kmoore<br/>
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4 Mark<br/>
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4 twilson<br/>
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3 may<br/>
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2 jrose<br/>
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2 kpfleming<br/>
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1 file<br/>
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1 jcolp<br/>
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1 Michael<br/>
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1 qwell<br/>
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</td>
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<td>
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2 Steve Davies<br/>
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2 Terry Wilson<br/>
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1 Dan Delaney<br/>
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1 Guenther Kelleter<br/>
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1 jamicque<br/>
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1 Julian Yap<br/>
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1 Michael L. Young<br/>
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1 Paul Belanger<br/>
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1 rmudgett<br/>
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1 Tilghman Lesher<br/>
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</td>
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<td>
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3 lmadsen<br/>
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2 fnordian<br/>
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2 one47<br/>
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1 alecdavis<br/>
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1 drdelaney<br/>
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1 elguero<br/>
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1 jamicque<br/>
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1 karlfife<br/>
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1 mdavenport<br/>
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1 mjordan<br/>
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1 mmichelson<br/>
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1 sdolloff<br/>
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1 themsley<br/>
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1 tomaso<br/>
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1 tsarik<br/>
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1 twilson<br/>
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1 vsauer<br/>
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</td>
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</tr>
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</table>
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<hr/>
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<a name="issues"><h2 align="center">Closed Issues</h2></a>
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<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>
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<h3>Category: Addons/chan_ooh323</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369090">369090</a><br/>
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Reporter: tsarik<br/>
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Coders: may<br/>
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<br/>
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<h3>Category: Applications/app_dial</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368898">368898</a><br/>
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Reporter: vsauer<br/>
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Coders: Mark<br/>
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<br/>
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<h3>Category: Applications/app_voicemail</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19923">ASTERISK-19923</a>: Asterisk crashing due to memory corruptions in chan_sip/voicemail<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369652">369652</a><br/>
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Reporter: drdelaney<br/>
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Testers: Dan Delaney, Julian Yap<br/>
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Coders: kmoore<br/>
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<br/>
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<h3>Category: Channels/chan_iax2</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19801">ASTERISK-19801</a>: Deadlock with masquerade and chan_iax<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368759">368759</a><br/>
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Reporter: alecdavis<br/>
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Testers: Guenther Kelleter<br/>
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Coders: rmudgett<br/>
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<br/>
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<h3>Category: Channels/chan_local</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368898">368898</a><br/>
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Reporter: vsauer<br/>
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Coders: Mark<br/>
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<br/>
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<h3>Category: Channels/chan_sip/General</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369090">369090</a><br/>
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Reporter: tsarik<br/>
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Coders: may<br/>
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<br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368807">368807</a><br/>
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Reporter: tomaso<br/>
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Coders: mmichelson<br/>
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<br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369352">369352</a><br/>
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Reporter: mmichelson<br/>
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Coders: mmichelson<br/>
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<br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369436">369436</a><br/>
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Reporter: one47<br/>
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Testers: Steve Davies, Terry Wilson<br/>
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Coders: twilson<br/>
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<br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369557">369557</a><br/>
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Reporter: one47<br/>
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Testers: Steve Davies, Terry Wilson<br/>
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Coders: twilson<br/>
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<br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20008">ASTERISK-20008</a>: outboundproxy ignored after when sending invite after 407<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369066">369066</a><br/>
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Reporter: fnordian<br/>
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Coders: Mark<br/>
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<br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20040">ASTERISK-20040</a>: Asterisk crashes when a guest call uses directmedia<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369214">369214</a><br/>
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Reporter: twilson<br/>
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Coders: twilson<br/>
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<br/>
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<h3>Category: Channels/chan_sip/IPv6</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369471">369471</a><br/>
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Reporter: lmadsen<br/>
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Coders: jcolp<br/>
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<br/>
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<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369302">369302</a><br/>
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Reporter: mjordan<br/>
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Coders: Mark<br/>
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<br/>
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<h3>Category: Core/Configuration</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19910">ASTERISK-19910</a>: Add sip_notify.conf entry for Digium phones<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369818">369818</a><br/>
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Reporter: mdavenport<br/>
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Coders: qwell<br/>
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<br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368873">368873</a><br/>
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Reporter: lmadsen<br/>
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Coders: mmichelson<br/>
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<br/>
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<h3>Category: Core/General</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369708">369708</a><br/>
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Reporter: fnordian<br/>
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Coders: mmichelson<br/>
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<br/>
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<h3>Category: Core/Netsock</h3><br/>
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<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20006">ASTERISK-20006</a>: Fix NULL pointer segfault in ast_sockaddr_parse()<br/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369108">369108</a><br/>
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Reporter: elguero<br/>
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Testers: Michael L. Young<br/>
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Coders: Michael<br/>
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<br/>
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<h3>Category: Documentation</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=369869">369869</a><br/>
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Reporter: lmadsen<br/>
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Coders: kmoore<br/>
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<br/>
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<h3>Category: General</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368830">368830</a><br/>
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Reporter: karlfife<br/>
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Testers: Paul Belanger, Tilghman Lesher<br/>
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Coders: mjordan<br/>
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<br/>
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<h3>Category: Resources/res_adsi</h3><br/>
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<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/>
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Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=368873">368873</a><br/>
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Reporter: lmadsen<br/>
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Coders: mmichelson<br/>
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<br/>
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<hr/>
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<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
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<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>
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<table width="100%" border="1">
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<td></td></tr></table>
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<hr/>
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<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
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<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>
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<pre>
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Makefile | 46 +--
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addons/chan_ooh323.c | 23 +
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addons/ooh323c/src/ooCalls.c | 3
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addons/ooh323c/src/ooq931.c | 2
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apps/app_dial.c | 34 +-
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apps/app_directory.c | 3
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apps/app_queue.c | 14 -
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apps/app_stack.c | 5
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apps/app_voicemail.c | 85 +++++-
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build_tools/find_missing_support_level | 3
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channels/chan_dahdi.c | 16 -
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channels/chan_iax2.c | 15 -
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channels/chan_misdn.c | 1
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channels/chan_sip.c | 233 +++++++++++++-----
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channels/console_board.c | 4
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channels/console_gui.c | 4
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channels/console_video.c | 4
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channels/iax2-parser.c | 4
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channels/iax2-provision.c | 4
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channels/misdn/ie.c | 4
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channels/misdn/isdn_lib.c | 4
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channels/misdn/isdn_msg_parser.c | 4
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channels/misdn/portinfo.c | 3
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channels/misdn_config.c | 4
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channels/sig_analog.c | 15 +
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channels/sig_pri.c | 3
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channels/sig_ss7.c | 3
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channels/sip/config_parser.c | 4
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channels/sip/dialplan_functions.c | 8
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channels/sip/include/sip.h | 4
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channels/sip/reqresp_parser.c | 6
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channels/sip/sdp_crypto.c | 8
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channels/sip/srtp.c | 4
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channels/vcodecs.c | 4
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channels/vgrabbers.c | 4
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configs/sip_notify.conf.sample | 5
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funcs/func_strings.c | 3
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funcs/func_volume.c | 3
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include/asterisk/adsi.h | 93 +++++--
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include/asterisk/channel.h | 2
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include/asterisk/netsock2.h | 3
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main/Makefile | 3
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main/abstract_jb.c | 4
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main/acl.c | 4
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main/adsi.c | 351 ++++++++++++++++++++++++++++
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main/alaw.c | 4
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main/aoc.c | 4
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main/app.c | 4
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main/asterisk.c | 4
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main/astfd.c | 4
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main/astmm.c | 4
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main/astobj2.c | 5
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main/audiohook.c | 4
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main/autochan.c | 4
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main/autoservice.c | 4
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main/bridging.c | 18 -
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main/callerid.c | 4
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main/ccss.c | 13 -
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main/cdr.c | 10
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main/cel.c | 4
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main/channel.c | 14 -
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main/chanvars.c | 4
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main/cli.c | 4
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main/config.c | 4
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main/data.c | 4
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main/datastore.c | 4
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main/db.c | 4
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main/devicestate.c | 4
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main/dial.c | 4
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main/dns.c | 4
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main/dnsmgr.c | 4
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main/dsp.c | 4
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main/enum.c | 4
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main/event.c | 4
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main/features.c | 409 ++++++++++++++++++---------------
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main/file.c | 4
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main/fixedjitterbuf.c | 4
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main/frame.c | 4
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main/framehook.c | 4
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main/fskmodem.c | 4
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main/fskmodem_float.c | 4
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main/fskmodem_int.c | 4
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main/global_datastores.c | 4
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main/hashtab.c | 4
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main/heap.c | 4
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main/image.c | 4
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main/indications.c | 4
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main/io.c | 4
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main/jitterbuf.c | 4
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main/loader.c | 8
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main/lock.c | 4
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main/logger.c | 4
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main/md5.c | 6
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main/netsock.c | 4
|
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main/netsock2.c | 10
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main/pbx.c | 24 +
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main/plc.c | 4
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main/privacy.c | 4
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main/rtp_engine.c | 4
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main/say.c | 6
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main/sched.c | 4
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main/security_events.c | 4
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main/slinfactory.c | 4
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main/srv.c | 4
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main/ssl.c | 4
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main/stdtime/localtime.c | 4
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main/strcompat.c | 4
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main/strings.c | 4
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main/stun.c | 4
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main/syslog.c | 4
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main/taskprocessor.c | 4
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main/tcptls.c | 7
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main/tdd.c | 4
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main/term.c | 4
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main/test.c | 4
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main/threadstorage.c | 4
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main/timing.c | 4
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main/translate.c | 4
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main/udptl.c | 7
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main/ulaw.c | 4
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main/utils.c | 4
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main/xml.c | 4
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main/xmldoc.c | 4
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pbx/dundi-parser.c | 4
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pbx/pbx_config.c | 4
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res/ael/pval.c | 4
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res/ais/clm.c | 4
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res/ais/evt.c | 4
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res/res_adsi.c | 187 ++++++++++-----
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res/res_adsi.exports.in | 33 --
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res/res_config_odbc.c | 7
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res/res_fax.c | 2
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res/res_odbc.c | 2
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res/res_smdi.c | 2
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res/res_speech.c | 3
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res/snmp/agent.c | 4
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136 files changed, 1597 insertions(+), 516 deletions(-)
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</pre><br/>
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<hr/>
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</body>
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</html>
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