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Creating tag for the release of asterisk-1.4.38-rc1

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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.38-rc1@295096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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1.4.38-rc1

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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-1.4.38-rc1</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-1.4.38-rc1</h3>
<h3 align="center">Date: 2010-11-15</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-1.4.37.</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>
5 jpeeler<br/>
3 rmudgett<br/>
1 alecdavis<br/>
1 dvossel<br/>
1 mnicholson<br/>
1 sruffell<br/>
1 tilghman<br/>
</td>
<td>
1 pabelanger<br/>
1 zerohalo<br/>
</td>
<td>
1 antonio<br/>
1 jmls<br/>
1 lftsy<br/>
1 netfuse<br/>
1 zerohalo<br/>
</td>
</tr>
</table>
<hr/>
<a name="issues"><h2 align="center">Closed Issues</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
<h3>Category: Applications/app_voicemail</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=18240">#18240</a>: [patch] VoicemailMain Exits Without Warning<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294903">294903</a><br/>
Reporter: netfuse<br/>
Testers: pabelanger<br/>
Coders: alecdavis<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17270">#17270</a>: getting warning message every 4 seconds<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294821">294821</a><br/>
Reporter: jmls<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16382">#16382</a>: [patch] SIP OPTIONS qualify message forever<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294688">294688</a><br/>
Reporter: lftsy<br/>
Testers: zerohalo<br/>
Coders: jpeeler<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=18123">#18123</a>: rtautoclear = no or rtautoclear = 0 not respected in sip.conf<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=293722">293722</a><br/>
Reporter: zerohalo<br/>
Coders: jpeeler<br/>
<br/>
<h3>Category: Core/PBX</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=18165">#18165</a>: [patch] hint state changes deadlock problem<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294384">294384</a><br/>
Reporter: antonio<br/>
Coders: jpeeler<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/1.4?view=revision&revision=293639">293639</a></td><td>rmudgett</td><td>Make warning message have more useful information in it.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=293805">293805</a></td><td>rmudgett</td><td>Party A in an analog 3-way call would continue to hear ringback after party C answers.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=293922">293922</a></td><td>dvossel</td><td>Fixes ringback tone on feature semi-attended transfer</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=293968">293968</a></td><td>sruffell</td><td>codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294163">294163</a></td><td>mnicholson</td><td>Modify our handling of 491 responses to drop any pending reinvite retry scheduler entries if we get a new 491.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294641">294641</a></td><td>jpeeler</td><td>One small addition to 294384 found while very carefully merging to 1.6.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=294739">294739</a></td><td>jpeeler</td><td>I didn't mean to merge this, sorry</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=295026">295026</a></td><td>tilghman</td><td>Create test verifying results of expression parser</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>
apps/app_voicemail.c | 3
channels/chan_dahdi.c | 48 +++++--
channels/chan_sip.c | 35 +++++
codecs/codec_dahdi.c | 8 -
include/asterisk.h | 1
main/asterisk.c | 1
main/pbx.c | 301 ++++++++++++++++++++++++++++++++------------------
res/res_features.c | 3
tests/test_expr.c | 191 +++++++++++++++++++++++++++++++
9 files changed, 461 insertions(+), 130 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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asterisk-1.4.38-rc1-summary.txt
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Release Summary
asterisk-1.4.38-rc1
Date: 2010-11-15
<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-1.4.37.
----------------------------------------------------------------------
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
5 jpeeler 1 pabelanger 1 antonio
3 rmudgett 1 zerohalo 1 jmls
1 alecdavis 1 lftsy
1 dvossel 1 netfuse
1 mnicholson 1 zerohalo
1 sruffell
1 tilghman
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Category: Applications/app_voicemail
#18240: [patch] VoicemailMain Exits Without Warning
Revision: 294903
Reporter: netfuse
Testers: pabelanger
Coders: alecdavis
Category: Channels/chan_dahdi
#17270: getting warning message every 4 seconds
Revision: 294821
Reporter: jmls
Coders: rmudgett
Category: Channels/chan_sip/General
#16382: [patch] SIP OPTIONS qualify message forever
Revision: 294688
Reporter: lftsy
Testers: zerohalo
Coders: jpeeler
#18123: rtautoclear = no or rtautoclear = 0 not respected in sip.conf
Revision: 293722
Reporter: zerohalo
Coders: jpeeler
Category: Core/PBX
#18165: [patch] hint state changes deadlock problem
Revision: 294384
Reporter: antonio
Coders: jpeeler
----------------------------------------------------------------------
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 |
|----------+------------+-----------------------------------+------------|
| 293639 | rmudgett | Make warning message have more | |
| | | useful information in it. | |
|----------+------------+-----------------------------------+------------|
| | | Party A in an analog 3-way call | |
| 293805 | rmudgett | would continue to hear ringback | |
| | | after party C answers. | |
|----------+------------+-----------------------------------+------------|
| 293922 | dvossel | Fixes ringback tone on feature | |
| | | semi-attended transfer | |
|----------+------------+-----------------------------------+------------|
| | | codecs/codec_dahdi: Prevent | |
| 293968 | sruffell | "choppy" audio when receiving | |
| | | unexpected frame sizes. | |
|----------+------------+-----------------------------------+------------|
| | | Modify our handling of 491 | |
| 294163 | mnicholson | responses to drop any pending | |
| | | reinvite retry scheduler entries | |
| | | if we get a new 491. | |
|----------+------------+-----------------------------------+------------|
| | | One small addition to 294384 | |
| 294641 | jpeeler | found while very carefully | |
| | | merging to 1.6. | |
|----------+------------+-----------------------------------+------------|
| 294739 | jpeeler | I didn't mean to merge this, | |
| | | sorry | |
|----------+------------+-----------------------------------+------------|
| 295026 | tilghman | Create test verifying results of | |
| | | expression parser | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
apps/app_voicemail.c | 3
channels/chan_dahdi.c | 48 +++++--
channels/chan_sip.c | 35 +++++
codecs/codec_dahdi.c | 8 -
include/asterisk.h | 1
main/asterisk.c | 1
main/pbx.c | 301 ++++++++++++++++++++++++++++++++------------------
res/res_features.c | 3
tests/test_expr.c | 191 +++++++++++++++++++++++++++++++
9 files changed, 461 insertions(+), 130 deletions(-)
----------------------------------------------------------------------

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tests/test_expr.c
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@@ -43,7 +43,7 @@ AST_TEST_DEFINE(expr_test)
{
int res = AST_TEST_PASS, i, len;
struct {
const char *input;
char *input;
const char *output;
} tests[] = {
{ "2 + 2", "4" },