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Author SHA1 Message Date
Leif Madsen
67eea66ffd Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 21:17:23 +00:00
Leif Madsen
da52af417c Importing release summary for 1.8.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 21:17:18 +00:00
Leif Madsen
c33755f47d Importing files for 1.8.0-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 21:17:16 +00:00
Leif Madsen
e489739e31 Creating tag for the release of asterisk-1.8.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 21:15:56 +00:00
Russell Bryant
a910d8db4f Creating tag for the release of asterisk-1.8.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 20:33:50 +00:00
Russell Bryant
3e0f9e93f6 Creating tag for the release of asterisk-1.8.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 20:26:06 +00:00
Leif Madsen
7c2ad640dd Creating tag for the release of asterisk-1.8.0-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.8.0-rc1@288489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22 19:29:39 +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-1.8.0-rc1</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-1.8.0-rc1</h3>
<h3 align="center">Date: 2010-09-22</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-1.8.0-beta5.</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>
23 tilghman<br/>
17 rmudgett<br/>
13 dvossel<br/>
8 russell<br/>
7 mnicholson<br/>
7 pitel<br/>
6 qwell<br/>
6 twilson<br/>
4 bbryant<br/>
2 alecdavis<br/>
2 jpeeler<br/>
2 oej<br/>
2 pabelanger<br/>
1 alexkuklin<br/>
1 andrew<br/>
1 avalentin<br/>
1 DEA<br/>
1 diruggles<br/>
1 junky<br/>
1 kobaz<br/>
1 kuj<br/>
1 makoto<br/>
1 mnick<br/>
1 Nick<br/>
1 pprindeville<br/>
1 sfritsch<br/>
1 sysreq<br/>
</td>
<td>
4 mnicholson<br/>
3 mkeuter<br/>
3 qwell<br/>
2 adriavidal<br/>
2 alecdavis<br/>
2 mich<br/>
2 ramonpeek<br/>
1 alexkuklin<br/>
1 andrew<br/>
1 avalentin<br/>
1 davidw<br/>
1 DEA<br/>
1 jamicque<br/>
1 jmhunter<br/>
1 junky<br/>
1 kuj<br/>
1 mdu113<br/>
1 Netview<br/>
1 Nick_Lewis<br/>
1 PavelL<br/>
1 pprindeville<br/>
1 ricardolandim<br/>
1 russell<br/>
1 schmidts<br/>
1 seanbright<br/>
1 sysreq<br/>
1 tilghman<br/>
1 twilson<br/>
</td>
<td>
4 oej<br/>
4 pitel<br/>
2 lmadsen<br/>
1 298<br/>
1 adriavidal<br/>
1 afried<br/>
1 Alexcr<br/>
1 alexkuklin<br/>
1 alexrecarey<br/>
1 amorsen<br/>
1 andrew<br/>
1 avalentin<br/>
1 Guggemand<br/>
1 haakon<br/>
1 ira<br/>
1 jamicque<br/>
1 jmhunter<br/>
1 jtodd<br/>
1 kobaz<br/>
1 kshumard<br/>
1 kuj<br/>
1 loloski<br/>
1 makoto<br/>
1 mdu113<br/>
1 mkeuter<br/>
1 Netview<br/>
1 Nick_Lewis<br/>
1 notthematrix<br/>
1 outcast<br/>
1 PavelL<br/>
1 pj<br/>
1 pprindeville<br/>
1 raarts<br/>
1 rain<br/>
1 ramonpeek<br/>
1 ricardolandim<br/>
1 russell<br/>
1 schmidts<br/>
1 seanbright<br/>
1 sysreq<br/>
1 tzafrir<br/>
1 under<br/>
1 vmikhnevych<br/>
1 wurstsalat<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/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17801">#17801</a>: [patch] ERROR[7169] astobj2.c: bad magic number 0x0 for 0x8b1c3d0<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287661">287661</a><br/>
Reporter: notthematrix<br/>
Testers: alecdavis<br/>
Coders: alecdavis<br/>
<br/>
<h3>Category: Applications/app_meetme</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17408">#17408</a>: [patch] MoH not restarted after end of conference announcement is played<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285533">285533</a><br/>
Reporter: sysreq<br/>
Testers: sysreq<br/>
Coders: sysreq<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17908">#17908</a>: [patch] MeetMe PIN handling broken<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287760">287760</a><br/>
Reporter: kuj<br/>
Testers: kuj<br/>
Coders: kuj<br/>
<br/>
<h3>Category: Applications/app_queue</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16893">#16893</a>: [patch] Realtime queue does not re-read announce variable from mysql after first use<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287388">287388</a><br/>
Reporter: haakon<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17535">#17535</a>: [patch] queue reload clears queue statistics<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284632">284632</a><br/>
Reporter: raarts<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17673">#17673</a>: [patch] When using Local/ as members, language is not inherited<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286189">286189</a><br/>
Reporter: Guggemand<br/>
Coders: twilson<br/>
<br/>
<h3>Category: Applications/app_voicemail</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=15726">#15726</a>: [patch] password change for mailboxes without user name<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285197">285197</a><br/>
Reporter: 298<br/>
Testers: junky<br/>
Coders: junky<br/>
<br/>
<h3>Category: CDR/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17900">#17900</a>: [patch] empty CDR variables and everything that goes after is not shown<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287116">287116</a><br/>
Reporter: under<br/>
Testers: mnicholson<br/>
Coders: mnicholson<br/>
<br/>
<h3>Category: CDR/cdr_pgsql</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16940">#16940</a>: [patch] Problem inserting CDR records when certain characters are used<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288268">288268</a><br/>
Reporter: jamicque<br/>
Testers: jamicque<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Channels/NewFeature</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17652">#17652</a>: [patch] Add CHANNEL(checkhangup) function<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285373">285373</a><br/>
Reporter: kobaz<br/>
Coders: kobaz<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16983">#16983</a>: alarm state not properly maintained on analog channels<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287683">287683</a><br/>
Reporter: tzafrir<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_iax2</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17919">#17919</a>: [patch] schedule_delivery calls ast_bridged_channel() on an unlocked channel<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288194">288194</a><br/>
Reporter: rain<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17935">#17935</a>: [patch] IAXregistry AMI does not return ActionID data<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284967">284967</a><br/>
Reporter: alexkuklin<br/>
Testers: alexkuklin<br/>
Coders: alexkuklin<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=18019">#18019</a>: [patch] chan_iax2 - timing interface missing<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288157">288157</a><br/>
Reporter: Netview<br/>
Testers: Netview<br/>
Coders: pabelanger<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17832">#17832</a>: [patch] SIP domains automatically add 0.0.0.0 and :: for IPv6<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285455">285455</a><br/>
Reporter: oej<br/>
Testers: qwell<br/>
Coders: qwell<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17960">#17960</a>: [patch] SIP peer wrong URI an to: tag<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286868">286868</a><br/>
Reporter: adriavidal<br/>
Testers: mich, mnicholson, adriavidal<br/>
Coders: mnicholson<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17981">#17981</a>: [patch] Wrong URI send if P-Assterted-Identiy is sent and caller is anonymous -> leads to reject on Aastra phone<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287893">287893</a><br/>
Reporter: avalentin<br/>
Testers: avalentin<br/>
Coders: avalentin<br/>
<br/>
<h3>Category: Channels/chan_sip/IPv6</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17831">#17831</a>: [patch] IPv6: SIp show settings doesn't show dual stack support<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285369">285369</a><br/>
Reporter: oej<br/>
Coders: qwell<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17840">#17840</a>: sip show settings: Internal IP with bindaddr=::<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286457">286457</a><br/>
Reporter: oej<br/>
Coders: qwell<br/>
<br/>
<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17005">#17005</a>: [patch] Asterisk sends session-timer with "require" after 15 minutes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285564">285564</a><br/>
Reporter: alexrecarey<br/>
Coders: dvossel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17676">#17676</a>: [patch] host not used in invite message, only the ip address.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286868">286868</a><br/>
Reporter: outcast<br/>
Testers: mich, mnicholson, adriavidal<br/>
Coders: mnicholson<br/>
<br/>
<h3>Category: Channels/chan_sip/Registration</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17551">#17551</a>: [patch] Realtime erase username when Unavailable<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286758">286758</a><br/>
Reporter: ricardolandim<br/>
Testers: ricardolandim, mnicholson<br/>
Coders: mnicholson<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=18017">#18017</a>: [patch] asterisk could not register to asterisk with pedantic=yes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287645">287645</a><br/>
Reporter: schmidts<br/>
Testers: schmidts<br/>
Coders: dvossel<br/>
<br/>
<h3>Category: Channels/chan_sip/Subscriptions</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17785">#17785</a>: [patch] Encoded URI in a subscription does not work<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288159">288159</a><br/>
Reporter: ramonpeek<br/>
Testers: ramonpeek<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17928">#17928</a>: [patch] AST_MAX_EXTENSION limitation on hint string length<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287120">287120</a><br/>
Reporter: mdu113<br/>
Testers: mdu113<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Channels/chan_skinny</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17680">#17680</a>: [patch] chan_skinny crashes asterisk when parking a call<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287643">287643</a><br/>
Reporter: jmhunter<br/>
Testers: jmhunter, DEA<br/>
Coders: DEA<br/>
<br/>
<h3>Category: Codecs/codec_gsm</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17688">#17688</a>: [patch] GCC 4.2.x optimizations result in improper behavior of GSM codec<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285819">285819</a><br/>
Reporter: pprindeville<br/>
Testers: mkeuter, pprindeville<br/>
Coders: pprindeville<br/>
<br/>
<h3>Category: Core/Channels</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17370">#17370</a>: [patch] ast_readstring (multiple DTMF input) doesn't transmit silence to the caller even if transmit_silence=yes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285745">285745</a><br/>
Reporter: makoto<br/>
Coders: makoto<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17794">#17794</a>: [patch] segfault on dialplan reload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285367">285367</a><br/>
Reporter: PavelL<br/>
Testers: PavelL<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Core/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16057">#16057</a>: [patch] Asterisk crashes with "Fixup failed on channel XXX, strange things may happen."<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287701">287701</a><br/>
Reporter: amorsen<br/>
Testers: ramonpeek, davidw, alecdavis<br/>
Coders: alecdavis<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17678">#17678</a>: Fix select() usage in Asterisk<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284597">284597</a><br/>
Reporter: russell<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17974">#17974</a>: VERBOSE message shows up on console when 'debug' enabled in logger.conf<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287639">287639</a><br/>
Reporter: lmadsen<br/>
Coders: bbryant<br/>
<br/>
<h3>Category: Core/ManagerInterface</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17891">#17891</a>: Possible memory leak in originate<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287471">287471</a><br/>
Reporter: oej<br/>
Coders: oej<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17917">#17917</a>: Reloads of manager.conf do not properly handle the resetting of options<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284781">284781</a><br/>
Reporter: lmadsen<br/>
Coders: bbryant<br/>
<br/>
<h3>Category: Core/PBX</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17902">#17902</a>: [patch] Asterisk 1.8.0-beta3 DNSMGR address corruption<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287895">287895</a><br/>
Reporter: afried<br/>
Testers: russell<br/>
Coders: russell<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17910">#17910</a>: Debian init script does not work<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287195">287195</a><br/>
Reporter: wurstsalat<br/>
Coders: qwell<br/>
<br/>
<h3>Category: Features/Parking</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=14882">#14882</a>: Parking extension number is not overriden in custom parking lots<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286931">286931</a><br/>
Reporter: vmikhnevych<br/>
Coders: mnick<br/>
<br/>
<h3>Category: Formats/format_wav</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=15029">#15029</a>: [patch] Add 16khz WAV support (format_wav16.c)<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284701">284701</a><br/>
Reporter: andrew<br/>
Testers: qwell, andrew<br/>
Coders: andrew<br/>
<br/>
<h3>Category: Functions/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17808">#17808</a>: [patch] Function CONNECTEDLINE causes Asterisk to exit<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284698">284698</a><br/>
Reporter: jtodd<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17707">#17707</a>: [patch] Upgrading from 1.6.2.10 to 1.8-beta1 did not work with the original modules.conf<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284610">284610</a><br/>
Reporter: ira<br/>
Testers: tilghman<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: PBX/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16903">#16903</a>: [patch] Incorrect pattern specificity in new dial pattern functions<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285711">285711</a><br/>
Reporter: Nick_Lewis<br/>
Testers: Nick_Lewis<br/>
Coders: Nick<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17882">#17882</a>: Crash in ast_frame_free<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288007">288007</a><br/>
Reporter: seanbright<br/>
Testers: seanbright<br/>
Coders: bbryant<br/>
<br/>
<h3>Category: PBX/pbx_spool</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17337">#17337</a>: [patch] [regression] flooding /var/spool/asterisk/outgoing/xxxxx.call: No such file or directory<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285386">285386</a><br/>
Reporter: loloski<br/>
Testers: mkeuter<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17924">#17924</a>: Call file errors in Asterisk 1.8beta<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285386">285386</a><br/>
Reporter: mkeuter<br/>
Testers: mkeuter<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Resources/res_calendar</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17776">#17776</a>: [patch] HTTP redirect support for calendars<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287269">287269</a><br/>
Reporter: pitel<br/>
Coders: pitel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17782">#17782</a>: [patch] If EWS request fails, asterisk crashes because of double free<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287270">287270</a><br/>
Reporter: pitel<br/>
Coders: pitel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17786">#17786</a>: [patch] Events are visible after they were removed from EWS calendar<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287271">287271</a><br/>
Reporter: pitel<br/>
Coders: pitel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17877">#17877</a>: [patch] Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286617">286617</a><br/>
Reporter: pitel<br/>
Coders: pitel<br/>
<br/>
<h3>Category: Resources/res_musiconhold</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16744">#16744</a>: [patch] 'moh reload' doesn't reload moh directory content<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285527">285527</a><br/>
Reporter: pj<br/>
Testers: qwell<br/>
Coders: qwell<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17807">#17807</a>: Music on hold doesn't recover very cleanly when it can't play a file<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285640">285640</a><br/>
Reporter: kshumard<br/>
Coders: bbryant<br/>
<br/>
<h3>Category: Resources/res_srtp</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17563">#17563</a>: [patch] SRTP (SRTP unprotect: authentication failure)<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287056">287056</a><br/>
Reporter: Alexcr<br/>
Testers: twilson<br/>
Coders: sfritsch<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=284561">284561</a></td><td>dvossel</td><td>During request to dialog matching, verify init_ruri is present before comparing.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284666">284666</a></td><td>tilghman</td><td>Fixing build.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284696">284696</a></td><td>tilghman</td><td>Fixing build</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284705">284705</a></td><td>dvossel</td><td>Removed relatedpeer code from sip_autodestruct</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284779">284779</a></td><td>rmudgett</td><td>Made output libpri event names if pri debugging is enabled when sig_pri processes them.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284780">284780</a></td><td>rmudgett</td><td>Simplified pri_dchannel() poll timeout duration code.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284849">284849</a></td><td>pitel</td><td>Support for calendar events priorities and categories</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284850">284850</a></td><td>pitel</td><td>Fix for calendar categories and priorities according to ISO C90</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284852">284852</a></td><td>pitel</td><td>Calendar categories and priorities: strdupa() fix</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284921">284921</a></td><td>twilson</td><td>Properly detect when a sound file doesn't exist</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284950">284950</a></td><td>dvossel</td><td>authenticate OPTIONS requests just like we would an INVITE</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=284952">284952</a></td><td>dvossel</td><td>During OPTIONS authentication, the authpeer does not need to be returned for any reason.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285006">285006</a></td><td>dvossel</td><td>Disables auth_options_request option by default.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285017">285017</a></td><td>twilson</td><td>Call correct lock function as transferer is a sip_pvt not a channel</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285057">285057</a></td><td>russell</td><td>Add a C++ compatible version of AST_CLI_DEFINE().</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285090">285090</a></td><td>tilghman</td><td>Silly convenience script for BSD platforms.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285161">285161</a></td><td>russell</td><td>Fix libsrtp -fPIC check for when non-standard prefix is used.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285162">285162</a></td><td>russell</td><td>regenerate configure script.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285195">285195</a></td><td>rmudgett</td><td></td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285268">285268</a></td><td>tilghman</td><td>Use poll, if indicated to do so, in the ast_poll2 implementation.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285336">285336</a></td><td>tilghman</td><td>Fix build on FreeBSD 8.0, take 2.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285371">285371</a></td><td>rmudgett</td><td>Fix cut-n-paste error.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285484">285484</a></td><td>tilghman</td><td>Documentation only</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285530">285530</a></td><td>qwell</td><td>Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285568">285568</a></td><td>dvossel</td><td>In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285931">285931</a></td><td>tilghman</td><td>Fix Mac OS X build.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285962">285962</a></td><td>tilghman</td><td>Another fix for Mac OS X.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=285992">285992</a></td><td>diruggles</td><td>Added missing documentation for ExternalIVR feature added in January 2010</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286025">286025</a></td><td>tilghman</td><td>Missing newline</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286112">286112</a></td><td>russell</td><td>Rate limit calls to fsync() to 1 per second after astdb updates.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286118">286118</a></td><td>rmudgett</td><td>An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286120">286120</a></td><td>pabelanger</td><td>Load iax.conf before registering any functions/applications/actions.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286270">286270</a></td><td>oej</td><td>Handle error response when we can't make file compatible</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286426">286426</a></td><td>rmudgett</td><td>Update chan_dahdi.conf.sample to reflect new libpri T309 default value.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286528">286528</a></td><td>tilghman</td><td>Refactor conversion to ast_poll() to fix callparking regression.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286558">286558</a></td><td>tilghman</td><td>C precedence got me</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286588">286588</a></td><td>tilghman</td><td>Add documentation on missing backend tables for Voicemail</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286647">286647</a></td><td>rmudgett</td><td>Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286682">286682</a></td><td>mnicholson</td><td>Only drop duplicate answer frames if the channel is bridged.</td>
<td><a href="https://issues.asterisk.org/view.php?id=2342">#2342</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286834">286834</a></td><td>dvossel</td><td>Sets subscribed type for outgoing MWI subscriptions so correct Event header is used.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286904">286904</a></td><td>rmudgett</td><td>Unable to originate calls using E&M over T1.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=286905">286905</a></td><td>rmudgett</td><td>Simplify some code in sig_analog.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287015">287015</a></td><td>jpeeler</td><td>Ensure mailbox is not filled to capacity before doing message forwarding.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287017">287017</a></td><td>rmudgett</td><td>Merged revision 287014 from</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287020">287020</a></td><td>jpeeler</td><td>fix uninintialized variable</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287193">287193</a></td><td>russell</td><td>Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287309">287309</a></td><td>mnicholson</td><td>Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().</td>
<td><a href="https://issues.asterisk.org/view.php?id=17928">#17928</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287559">287559</a></td><td>mnicholson</td><td>Use ast_str when processing hint state changes</td>
<td><a href="https://issues.asterisk.org/view.php?id=17928">#17928</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287647">287647</a></td><td>dvossel</td><td>Addition of the FrameHook API (AKA AwesomeHooks)</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287757">287757</a></td><td>twilson</td><td>Avoid infinite loop with certain local channel connected line updates</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287833">287833</a></td><td>twilson</td><td>Don't generate connected line buffer twice for comparison</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287863">287863</a></td><td>russell</td><td>Fix a regression in verbose logger processing.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287897">287897</a></td><td>rmudgett</td><td>Cut-n-paste error in builtin_blindtransfer().</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287929">287929</a></td><td>dvossel</td><td>Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287931">287931</a></td><td>twilson</td><td>Revert change in favor of a more targeted fix</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=287935">287935</a></td><td>tilghman</td><td>Less than zero is an error, not any non-zero value.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288079">288079</a></td><td>rmudgett</td><td>Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288080">288080</a></td><td>rmudgett</td><td>Simplify locking code for REDIRECTING interception macro when forwarding a call.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288082">288082</a></td><td>rmudgett</td><td>Add note in party manipulation chapter on interception macros.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288341">288341</a></td><td>russell</td><td>Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288345">288345</a></td><td>dvossel</td><td>During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.8?view=revision&revision=288418">288418</a></td><td>dvossel</td><td>RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.</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>
BSDmakefile | 11
CHANGES | 8
UPGRADE.txt | 8
addons/ooh323c/src/ooSocket.h | 5
addons/ooh323c/src/oochannels.c | 24 -
apps/app_adsiprog.c | 6
apps/app_chanspy.c | 2
apps/app_dial.c | 16
apps/app_followme.c | 1
apps/app_getcpeid.c | 6
apps/app_meetme.c | 24 +
apps/app_queue.c | 34 -
apps/app_speech_utils.c | 6
apps/app_stack.c | 6
apps/app_voicemail.c | 47 ++
cdr/cdr_pgsql.c | 21 +
channels/chan_agent.c | 2
channels/chan_dahdi.c | 58 ++-
channels/chan_iax2.c | 397 +++++++++++-----------
channels/chan_local.c | 67 +++
channels/chan_mgcp.c | 5
channels/chan_misdn.c | 257 ++++++++++----
channels/chan_phone.c | 80 ++--
channels/chan_sip.c | 233 ++++++++----
channels/chan_skinny.c | 28 +
channels/chan_usbradio.c | 159 ++++----
channels/console_video.c | 233 ++++++------
channels/misdn/isdn_msg_parser.c | 66 +++
channels/sig_analog.c | 51 +-
channels/sig_pri.c | 68 ++-
channels/sig_pri.h | 2
channels/sip/include/sip.h | 3
codecs/gsm/Makefile | 8
configs/cdr_pgsql.conf.sample | 1
configs/chan_dahdi.conf.sample | 3
configs/features.conf.sample | 3
configs/queues.conf.sample | 15
configs/sip.conf.sample | 3
configure.ac | 69 +++
contrib/init.d/rc.debian.asterisk | 2
contrib/realtime/mysql/voicemail_data.sql | 29 +
contrib/realtime/mysql/voicemail_messages.sql | 29 +
doc/externalivr.txt | 14
doc/tex/asterisk.tex | 3
doc/tex/channelvariables.tex | 23 +
doc/tex/partymanip.tex | 331 ++++++++++++++++++
formats/format_wav.c | 69 ++-
funcs/func_aes.c | 6
funcs/func_channel.c | 29 +
funcs/func_frame_trace.c | 365 ++++++++++++++++++++
include/asterisk/astobj2.h | 4
include/asterisk/autoconfig.h.in | 74 ++--
include/asterisk/calendar.h | 2
include/asterisk/channel.h | 61 +--
include/asterisk/cli.h | 4
include/asterisk/compiler.h | 6
include/asterisk/features.h | 10
include/asterisk/frame.h | 7
include/asterisk/framehook.h | 311 +++++++++++++++++
include/asterisk/module.h | 5
include/asterisk/pbx.h | 7
include/asterisk/poll-compat.h | 22 +
include/asterisk/select.h | 109 ++++++
main/acl.c | 18 -
main/asterisk.c | 48 ++
main/cdr.c | 20 -
main/channel.c | 118 +++++-
main/db.c | 51 ++
main/dnsmgr.c | 19 -
main/features.c | 467 +++++++++++++++-----------
main/file.c | 5
main/framehook.c | 184 ++++++++++
main/loader.c | 20 +
main/logger.c | 74 +---
main/manager.c | 36 +-
main/pbx.c | 29 +
main/poll.c | 196 +++++-----
main/stun.c | 7
makeopts.in | 1
pbx/pbx_config.c | 28 +
pbx/pbx_dundi.c | 1
pbx/pbx_loopback.c | 1
pbx/pbx_realtime.c | 1
pbx/pbx_spool.c | 15
res/res_ais.c | 20 -
res/res_calendar.c | 22 +
res/res_calendar_caldav.c | 16
res/res_calendar_ews.c | 103 +++++
res/res_calendar_exchange.c | 2
res/res_calendar_icalendar.c | 10
res/res_jabber.c | 23 -
res/res_musiconhold.c | 25 +
res/res_pktccops.c | 35 +
res/res_rtp_asterisk.c | 7
res/res_srtp.c | 5
tests/test_poll.c | 247 +++++++++++++
96 files changed, 4119 insertions(+), 1293 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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@@ -0,0 +1,790 @@
Release Summary
asterisk-1.8.0-rc1
Date: 2010-09-22
<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-1.8.0-beta5.
----------------------------------------------------------------------
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
23 tilghman 4 mnicholson 4 oej
17 rmudgett 3 mkeuter 4 pitel
13 dvossel 3 qwell 2 lmadsen
8 russell 2 adriavidal 1 298
7 mnicholson 2 alecdavis 1 adriavidal
7 pitel 2 mich 1 afried
6 qwell 2 ramonpeek 1 Alexcr
6 twilson 1 alexkuklin 1 alexkuklin
4 bbryant 1 andrew 1 alexrecarey
2 alecdavis 1 avalentin 1 amorsen
2 jpeeler 1 davidw 1 andrew
2 oej 1 DEA 1 avalentin
2 pabelanger 1 jamicque 1 Guggemand
1 alexkuklin 1 jmhunter 1 haakon
1 andrew 1 junky 1 ira
1 avalentin 1 kuj 1 jamicque
1 DEA 1 mdu113 1 jmhunter
1 diruggles 1 Netview 1 jtodd
1 junky 1 Nick_Lewis 1 kobaz
1 kobaz 1 PavelL 1 kshumard
1 kuj 1 pprindeville 1 kuj
1 makoto 1 ricardolandim 1 loloski
1 mnick 1 russell 1 makoto
1 Nick 1 schmidts 1 mdu113
1 pprindeville 1 seanbright 1 mkeuter
1 sfritsch 1 sysreq 1 Netview
1 sysreq 1 tilghman 1 Nick_Lewis
1 twilson 1 notthematrix
1 outcast
1 PavelL
1 pj
1 pprindeville
1 raarts
1 rain
1 ramonpeek
1 ricardolandim
1 russell
1 schmidts
1 seanbright
1 sysreq
1 tzafrir
1 under
1 vmikhnevych
1 wurstsalat
----------------------------------------------------------------------
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/General
#17801: [patch] ERROR[7169] astobj2.c: bad magic number 0x0 for 0x8b1c3d0
Revision: 287661
Reporter: notthematrix
Testers: alecdavis
Coders: alecdavis
Category: Applications/app_meetme
#17408: [patch] MoH not restarted after end of conference announcement is
played
Revision: 285533
Reporter: sysreq
Testers: sysreq
Coders: sysreq
#17908: [patch] MeetMe PIN handling broken
Revision: 287760
Reporter: kuj
Testers: kuj
Coders: kuj
Category: Applications/app_queue
#16893: [patch] Realtime queue does not re-read announce variable from
mysql after first use
Revision: 287388
Reporter: haakon
Coders: tilghman
#17535: [patch] queue reload clears queue statistics
Revision: 284632
Reporter: raarts
Coders: tilghman
#17673: [patch] When using Local/ as members, language is not inherited
Revision: 286189
Reporter: Guggemand
Coders: twilson
Category: Applications/app_voicemail
#15726: [patch] password change for mailboxes without user name
Revision: 285197
Reporter: 298
Testers: junky
Coders: junky
Category: CDR/General
#17900: [patch] empty CDR variables and everything that goes after is not
shown
Revision: 287116
Reporter: under
Testers: mnicholson
Coders: mnicholson
Category: CDR/cdr_pgsql
#16940: [patch] Problem inserting CDR records when certain characters are
used
Revision: 288268
Reporter: jamicque
Testers: jamicque
Coders: tilghman
Category: Channels/NewFeature
#17652: [patch] Add CHANNEL(checkhangup) function
Revision: 285373
Reporter: kobaz
Coders: kobaz
Category: Channels/chan_dahdi
#16983: alarm state not properly maintained on analog channels
Revision: 287683
Reporter: tzafrir
Coders: rmudgett
Category: Channels/chan_iax2
#17919: [patch] schedule_delivery calls ast_bridged_channel() on an
unlocked channel
Revision: 288194
Reporter: rain
Coders: rmudgett
#17935: [patch] IAXregistry AMI does not return ActionID data
Revision: 284967
Reporter: alexkuklin
Testers: alexkuklin
Coders: alexkuklin
#18019: [patch] chan_iax2 - timing interface missing
Revision: 288157
Reporter: Netview
Testers: Netview
Coders: pabelanger
Category: Channels/chan_sip/General
#17832: [patch] SIP domains automatically add 0.0.0.0 and :: for IPv6
Revision: 285455
Reporter: oej
Testers: qwell
Coders: qwell
#17960: [patch] SIP peer wrong URI an to: tag
Revision: 286868
Reporter: adriavidal
Testers: mich, mnicholson, adriavidal
Coders: mnicholson
#17981: [patch] Wrong URI send if P-Assterted-Identiy is sent and caller
is anonymous -> leads to reject on Aastra phone
Revision: 287893
Reporter: avalentin
Testers: avalentin
Coders: avalentin
Category: Channels/chan_sip/IPv6
#17831: [patch] IPv6: SIp show settings doesn't show dual stack support
Revision: 285369
Reporter: oej
Coders: qwell
#17840: sip show settings: Internal IP with bindaddr=::
Revision: 286457
Reporter: oej
Coders: qwell
Category: Channels/chan_sip/Interoperability
#17005: [patch] Asterisk sends session-timer with "require" after 15
minutes
Revision: 285564
Reporter: alexrecarey
Coders: dvossel
#17676: [patch] host not used in invite message, only the ip address.
Revision: 286868
Reporter: outcast
Testers: mich, mnicholson, adriavidal
Coders: mnicholson
Category: Channels/chan_sip/Registration
#17551: [patch] Realtime erase username when Unavailable
Revision: 286758
Reporter: ricardolandim
Testers: ricardolandim, mnicholson
Coders: mnicholson
#18017: [patch] asterisk could not register to asterisk with pedantic=yes
Revision: 287645
Reporter: schmidts
Testers: schmidts
Coders: dvossel
Category: Channels/chan_sip/Subscriptions
#17785: [patch] Encoded URI in a subscription does not work
Revision: 288159
Reporter: ramonpeek
Testers: ramonpeek
Coders: tilghman
#17928: [patch] AST_MAX_EXTENSION limitation on hint string length
Revision: 287120
Reporter: mdu113
Testers: mdu113
Coders: tilghman
Category: Channels/chan_skinny
#17680: [patch] chan_skinny crashes asterisk when parking a call
Revision: 287643
Reporter: jmhunter
Testers: jmhunter, DEA
Coders: DEA
Category: Codecs/codec_gsm
#17688: [patch] GCC 4.2.x optimizations result in improper behavior of GSM
codec
Revision: 285819
Reporter: pprindeville
Testers: mkeuter, pprindeville
Coders: pprindeville
Category: Core/Channels
#17370: [patch] ast_readstring (multiple DTMF input) doesn't transmit
silence to the caller even if transmit_silence=yes
Revision: 285745
Reporter: makoto
Coders: makoto
Category: Core/Configuration
#17794: [patch] segfault on dialplan reload
Revision: 285367
Reporter: PavelL
Testers: PavelL
Coders: tilghman
Category: Core/General
#16057: [patch] Asterisk crashes with "Fixup failed on channel XXX,
strange things may happen."
Revision: 287701
Reporter: amorsen
Testers: ramonpeek, davidw, alecdavis
Coders: alecdavis
#17678: Fix select() usage in Asterisk
Revision: 284597
Reporter: russell
Coders: tilghman
#17974: VERBOSE message shows up on console when 'debug' enabled in
logger.conf
Revision: 287639
Reporter: lmadsen
Coders: bbryant
Category: Core/ManagerInterface
#17891: Possible memory leak in originate
Revision: 287471
Reporter: oej
Coders: oej
#17917: Reloads of manager.conf do not properly handle the resetting of
options
Revision: 284781
Reporter: lmadsen
Coders: bbryant
Category: Core/PBX
#17902: [patch] Asterisk 1.8.0-beta3 DNSMGR address corruption
Revision: 287895
Reporter: afried
Testers: russell
Coders: russell
#17910: Debian init script does not work
Revision: 287195
Reporter: wurstsalat
Coders: qwell
Category: Features/Parking
#14882: Parking extension number is not overriden in custom parking lots
Revision: 286931
Reporter: vmikhnevych
Coders: mnick
Category: Formats/format_wav
#15029: [patch] Add 16khz WAV support (format_wav16.c)
Revision: 284701
Reporter: andrew
Testers: qwell, andrew
Coders: andrew
Category: Functions/General
#17808: [patch] Function CONNECTEDLINE causes Asterisk to exit
Revision: 284698
Reporter: jtodd
Coders: rmudgett
Category: General
#17707: [patch] Upgrading from 1.6.2.10 to 1.8-beta1 did not work with the
original modules.conf
Revision: 284610
Reporter: ira
Testers: tilghman
Coders: tilghman
Category: PBX/General
#16903: [patch] Incorrect pattern specificity in new dial pattern
functions
Revision: 285711
Reporter: Nick_Lewis
Testers: Nick_Lewis
Coders: Nick
#17882: Crash in ast_frame_free
Revision: 288007
Reporter: seanbright
Testers: seanbright
Coders: bbryant
Category: PBX/pbx_spool
#17337: [patch] [regression] flooding
/var/spool/asterisk/outgoing/xxxxx.call: No such file or directory
Revision: 285386
Reporter: loloski
Testers: mkeuter
Coders: tilghman
#17924: Call file errors in Asterisk 1.8beta
Revision: 285386
Reporter: mkeuter
Testers: mkeuter
Coders: tilghman
Category: Resources/res_calendar
#17776: [patch] HTTP redirect support for calendars
Revision: 287269
Reporter: pitel
Coders: pitel
#17782: [patch] If EWS request fails, asterisk crashes because of double
free
Revision: 287270
Reporter: pitel
Coders: pitel
#17786: [patch] Events are visible after they were removed from EWS
calendar
Revision: 287271
Reporter: pitel
Coders: pitel
#17877: [patch] Merging events for Exchange web service doesn't work as
expected, resulting in only one event in calendar
Revision: 286617
Reporter: pitel
Coders: pitel
Category: Resources/res_musiconhold
#16744: [patch] 'moh reload' doesn't reload moh directory content
Revision: 285527
Reporter: pj
Testers: qwell
Coders: qwell
#17807: Music on hold doesn't recover very cleanly when it can't play a
file
Revision: 285640
Reporter: kshumard
Coders: bbryant
Category: Resources/res_srtp
#17563: [patch] SRTP (SRTP unprotect: authentication failure)
Revision: 287056
Reporter: Alexcr
Testers: twilson
Coders: sfritsch
----------------------------------------------------------------------
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 |
|----------+------------+-----------------------------------+------------|
| | | During request to dialog | |
| 284561 | dvossel | matching, verify init_ruri is | |
| | | present before comparing. | |
|----------+------------+-----------------------------------+------------|
| 284666 | tilghman | Fixing build. | |
|----------+------------+-----------------------------------+------------|
| 284696 | tilghman | Fixing build | |
|----------+------------+-----------------------------------+------------|
| 284705 | dvossel | Removed relatedpeer code from | |
| | | sip_autodestruct | |
|----------+------------+-----------------------------------+------------|
| | | Made output libpri event names if | |
| 284779 | rmudgett | pri debugging is enabled when | |
| | | sig_pri processes them. | |
|----------+------------+-----------------------------------+------------|
| 284780 | rmudgett | Simplified pri_dchannel() poll | |
| | | timeout duration code. | |
|----------+------------+-----------------------------------+------------|
| 284849 | pitel | Support for calendar events | |
| | | priorities and categories | |
|----------+------------+-----------------------------------+------------|
| 284850 | pitel | Fix for calendar categories and | |
| | | priorities according to ISO C90 | |
|----------+------------+-----------------------------------+------------|
| 284852 | pitel | Calendar categories and | |
| | | priorities: strdupa() fix | |
|----------+------------+-----------------------------------+------------|
| 284921 | twilson | Properly detect when a sound file | |
| | | doesn't exist | |
|----------+------------+-----------------------------------+------------|
| 284950 | dvossel | authenticate OPTIONS requests | |
| | | just like we would an INVITE | |
|----------+------------+-----------------------------------+------------|
| | | During OPTIONS authentication, | |
| 284952 | dvossel | the authpeer does not need to be | |
| | | returned for any reason. | |
|----------+------------+-----------------------------------+------------|
| 285006 | dvossel | Disables auth_options_request | |
| | | option by default. | |
|----------+------------+-----------------------------------+------------|
| | | Call correct lock function as | |
| 285017 | twilson | transferer is a sip_pvt not a | |
| | | channel | |
|----------+------------+-----------------------------------+------------|
| 285057 | russell | Add a C++ compatible version of | |
| | | AST_CLI_DEFINE(). | |
|----------+------------+-----------------------------------+------------|
| 285090 | tilghman | Silly convenience script for BSD | |
| | | platforms. | |
|----------+------------+-----------------------------------+------------|
| 285161 | russell | Fix libsrtp -fPIC check for when | |
| | | non-standard prefix is used. | |
|----------+------------+-----------------------------------+------------|
| 285162 | russell | regenerate configure script. | |
|----------+------------+-----------------------------------+------------|
| 285195 | rmudgett | | |
|----------+------------+-----------------------------------+------------|
| 285268 | tilghman | Use poll, if indicated to do so, | |
| | | in the ast_poll2 implementation. | |
|----------+------------+-----------------------------------+------------|
| 285336 | tilghman | Fix build on FreeBSD 8.0, take 2. | |
|----------+------------+-----------------------------------+------------|
| 285371 | rmudgett | Fix cut-n-paste error. | |
|----------+------------+-----------------------------------+------------|
| 285484 | tilghman | Documentation only | |
|----------+------------+-----------------------------------+------------|
| | | Follow coding guidelines in moh | |
| 285530 | qwell | rescan fix. Also fix the | |
| | | documentation that got me in | |
| | | trouble. | |
|----------+------------+-----------------------------------+------------|
| | | In retrans_pkt, do not unlock pvt | |
| 285568 | dvossel | until the end of the function on | |
| | | a transmit failure. | |
|----------+------------+-----------------------------------+------------|
| 285931 | tilghman | Fix Mac OS X build. | |
|----------+------------+-----------------------------------+------------|
| 285962 | tilghman | Another fix for Mac OS X. | |
|----------+------------+-----------------------------------+------------|
| | | Added missing documentation for | |
| 285992 | diruggles | ExternalIVR feature added in | |
| | | January 2010 | |
|----------+------------+-----------------------------------+------------|
| 286025 | tilghman | Missing newline | |
|----------+------------+-----------------------------------+------------|
| 286112 | russell | Rate limit calls to fsync() to 1 | |
| | | per second after astdb updates. | |
|----------+------------+-----------------------------------+------------|
| | | An outgoing call may not get hung | |
| 286118 | rmudgett | up if a pre-connect incoming ISDN | |
| | | call is disconnected. | |
|----------+------------+-----------------------------------+------------|
| | | Load iax.conf before registering | |
| 286120 | pabelanger | any | |
| | | functions/applications/actions. | |
|----------+------------+-----------------------------------+------------|
| 286270 | oej | Handle error response when we | |
| | | can't make file compatible | |
|----------+------------+-----------------------------------+------------|
| | | Update chan_dahdi.conf.sample to | |
| 286426 | rmudgett | reflect new libpri T309 default | |
| | | value. | |
|----------+------------+-----------------------------------+------------|
| 286528 | tilghman | Refactor conversion to ast_poll() | |
| | | to fix callparking regression. | |
|----------+------------+-----------------------------------+------------|
| 286558 | tilghman | C precedence got me | |
|----------+------------+-----------------------------------+------------|
| 286588 | tilghman | Add documentation on missing | |
| | | backend tables for Voicemail | |
|----------+------------+-----------------------------------+------------|
| | | Corrected documented | |
| 286647 | rmudgett | CONNECTED_LINE and REDIRECTING | |
| | | party manipulation macro names. | |
|----------+------------+-----------------------------------+------------|
| 286682 | mnicholson | Only drop duplicate answer frames | #2342 |
| | | if the channel is bridged. | |
|----------+------------+-----------------------------------+------------|
| | | Sets subscribed type for outgoing | |
| 286834 | dvossel | MWI subscriptions so correct | |
| | | Event header is used. | |
|----------+------------+-----------------------------------+------------|
| 286904 | rmudgett | Unable to originate calls using | |
| | | E&M over T1. | |
|----------+------------+-----------------------------------+------------|
| 286905 | rmudgett | Simplify some code in sig_analog. | |
|----------+------------+-----------------------------------+------------|
| | | Ensure mailbox is not filled to | |
| 287015 | jpeeler | capacity before doing message | |
| | | forwarding. | |
|----------+------------+-----------------------------------+------------|
| 287017 | rmudgett | Merged revision 287014 from | |
|----------+------------+-----------------------------------+------------|
| 287020 | jpeeler | fix uninintialized variable | |
|----------+------------+-----------------------------------+------------|
| | | Set the default for "autofill" | |
| 287193 | russell | and "shared_lastcall" to "yes" in | |
| | | queues.conf. | |
|----------+------------+-----------------------------------+------------|
| | | Use ast_strdup() instead of | |
| 287309 | mnicholson | ast_strdupa() while processing in | #17928 |
| | | ast_hint_state_changed(). | |
|----------+------------+-----------------------------------+------------|
| 287559 | mnicholson | Use ast_str when processing hint | #17928 |
| | | state changes | |
|----------+------------+-----------------------------------+------------|
| 287647 | dvossel | Addition of the FrameHook API | |
| | | (AKA AwesomeHooks) | |
|----------+------------+-----------------------------------+------------|
| | | Avoid infinite loop with certain | |
| 287757 | twilson | local channel connected line | |
| | | updates | |
|----------+------------+-----------------------------------+------------|
| 287833 | twilson | Don't generate connected line | |
| | | buffer twice for comparison | |
|----------+------------+-----------------------------------+------------|
| 287863 | russell | Fix a regression in verbose | |
| | | logger processing. | |
|----------+------------+-----------------------------------+------------|
| 287897 | rmudgett | Cut-n-paste error in | |
| | | builtin_blindtransfer(). | |
|----------+------------+-----------------------------------+------------|
| | | Send a "415 Unsupported Media | |
| 287929 | dvossel | Type" after failure to process | |
| | | sdp due to unknown | |
| | | Content-Encoding header. | |
|----------+------------+-----------------------------------+------------|
| 287931 | twilson | Revert change in favor of a more | |
| | | targeted fix | |
|----------+------------+-----------------------------------+------------|
| 287935 | tilghman | Less than zero is an error, not | |
| | | any non-zero value. | |
|----------+------------+-----------------------------------+------------|
| | | Protect channel access in | |
| 288079 | rmudgett | CONNECTED_LINE and REDIRECTING | |
| | | interception macro launch code. | |
|----------+------------+-----------------------------------+------------|
| | | Simplify locking code for | |
| 288080 | rmudgett | REDIRECTING interception macro | |
| | | when forwarding a call. | |
|----------+------------+-----------------------------------+------------|
| 288082 | rmudgett | Add note in party manipulation | |
| | | chapter on interception macros. | |
|----------+------------+-----------------------------------+------------|
| | | Fix a 100% CPU consumption | |
| 288341 | russell | problem when setting console=yes | |
| | | in asterisk.conf. | |
|----------+------------+-----------------------------------+------------|
| | | During check_pendings, if the | |
| 288345 | dvossel | dialog is terminated with a | |
| | | CANCEL, change the invitestate to | |
| | | INV_CANCEL like in sip_hangup. | |
|----------+------------+-----------------------------------+------------|
| | | RFC3261 section 12.2 explicitly | |
| 288418 | dvossel | says out of order requests are | |
| | | responded with a 500 Server | |
| | | Internal Error response. | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
BSDmakefile | 11
CHANGES | 8
UPGRADE.txt | 8
addons/ooh323c/src/ooSocket.h | 5
addons/ooh323c/src/oochannels.c | 24 -
apps/app_adsiprog.c | 6
apps/app_chanspy.c | 2
apps/app_dial.c | 16
apps/app_followme.c | 1
apps/app_getcpeid.c | 6
apps/app_meetme.c | 24 +
apps/app_queue.c | 34 -
apps/app_speech_utils.c | 6
apps/app_stack.c | 6
apps/app_voicemail.c | 47 ++
cdr/cdr_pgsql.c | 21 +
channels/chan_agent.c | 2
channels/chan_dahdi.c | 58 ++-
channels/chan_iax2.c | 397 +++++++++++-----------
channels/chan_local.c | 67 +++
channels/chan_mgcp.c | 5
channels/chan_misdn.c | 257 ++++++++++----
channels/chan_phone.c | 80 ++--
channels/chan_sip.c | 233 ++++++++----
channels/chan_skinny.c | 28 +
channels/chan_usbradio.c | 159 ++++----
channels/console_video.c | 233 ++++++------
channels/misdn/isdn_msg_parser.c | 66 +++
channels/sig_analog.c | 51 +-
channels/sig_pri.c | 68 ++-
channels/sig_pri.h | 2
channels/sip/include/sip.h | 3
codecs/gsm/Makefile | 8
configs/cdr_pgsql.conf.sample | 1
configs/chan_dahdi.conf.sample | 3
configs/features.conf.sample | 3
configs/queues.conf.sample | 15
configs/sip.conf.sample | 3
configure.ac | 69 +++
contrib/init.d/rc.debian.asterisk | 2
contrib/realtime/mysql/voicemail_data.sql | 29 +
contrib/realtime/mysql/voicemail_messages.sql | 29 +
doc/externalivr.txt | 14
doc/tex/asterisk.tex | 3
doc/tex/channelvariables.tex | 23 +
doc/tex/partymanip.tex | 331 ++++++++++++++++++
formats/format_wav.c | 69 ++-
funcs/func_aes.c | 6
funcs/func_channel.c | 29 +
funcs/func_frame_trace.c | 365 ++++++++++++++++++++
include/asterisk/astobj2.h | 4
include/asterisk/autoconfig.h.in | 74 ++--
include/asterisk/calendar.h | 2
include/asterisk/channel.h | 61 +--
include/asterisk/cli.h | 4
include/asterisk/compiler.h | 6
include/asterisk/features.h | 10
include/asterisk/frame.h | 7
include/asterisk/framehook.h | 311 +++++++++++++++++
include/asterisk/module.h | 5
include/asterisk/pbx.h | 7
include/asterisk/poll-compat.h | 22 +
include/asterisk/select.h | 109 ++++++
main/acl.c | 18 -
main/asterisk.c | 48 ++
main/cdr.c | 20 -
main/channel.c | 118 +++++-
main/db.c | 51 ++
main/dnsmgr.c | 19 -
main/features.c | 467 +++++++++++++++-----------
main/file.c | 5
main/framehook.c | 184 ++++++++++
main/loader.c | 20 +
main/logger.c | 74 +---
main/manager.c | 36 +-
main/pbx.c | 29 +
main/poll.c | 196 +++++-----
main/stun.c | 7
makeopts.in | 1
pbx/pbx_config.c | 28 +
pbx/pbx_dundi.c | 1
pbx/pbx_loopback.c | 1
pbx/pbx_realtime.c | 1
pbx/pbx_spool.c | 15
res/res_ais.c | 20 -
res/res_calendar.c | 22 +
res/res_calendar_caldav.c | 16
res/res_calendar_ews.c | 103 +++++
res/res_calendar_exchange.c | 2
res/res_calendar_icalendar.c | 10
res/res_jabber.c | 23 -
res/res_musiconhold.c | 25 +
res/res_pktccops.c | 35 +
res/res_rtp_asterisk.c | 7
res/res_srtp.c | 5
tests/test_poll.c | 247 +++++++++++++
96 files changed, 4119 insertions(+), 1293 deletions(-)
----------------------------------------------------------------------