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Leif Madsen
6929baf5e7 Importing release summary for 1.4.36 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 20:28:07 +00:00
Leif Madsen
42eb1cfdc0 Oops, minor typo causes these to be innaccurate. Will rebuild.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 20:27:12 +00:00
Leif Madsen
d95f959e64 Importing release summary for 1.4.36 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 20:06:00 +00:00
Leif Madsen
25cbca3c72 Remove release summaries as we'll rebuild them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 20:04:53 +00:00
Leif Madsen
3f6741de5c Update .version and ChangeLog
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 20:00:21 +00:00
Leif Madsen
180fdd827b Create Asterisk 1.4.36 from 1.4.36-rc1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36@286459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13 19:51:50 +00:00
Leif Madsen
a7ef5261f2 Use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36-rc1@283276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23 18:24:11 +00:00
Leif Madsen
72e8f11d85 Importing release summary for 1.4.36-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36-rc1@283275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23 18:24:06 +00:00
Leif Madsen
f732ffcbec Importing files for 1.4.36-rc1 release.
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36-rc1@283274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23 18:24:03 +00:00
Leif Madsen
16e3d6d713 Creating tag for the release of asterisk-1.4.36-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36-rc1@283273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23 18:22:00 +00:00
Leif Madsen
8b18237587 Creating tag for the release of asterisk-1.4.36-rc1
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.36-rc1@283271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-23 18:21:12 +00:00
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-1.4.36</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-1.4.36</h3>
<h3 align="center">Date: 2010-09-13</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-1.4.35.</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>
4 dvossel<br/>
4 tilghman<br/>
3 jpeeler<br/>
3 rmudgett<br/>
2 RoadKill<br/>
2 twilson<br/>
1 addix<br/>
1 jeang<br/>
1 klaus3000<br/>
1 lmadsen<br/>
1 mmichelson<br/>
1 nic<br/>
1 qwell<br/>
1 russell<br/>
1 snuffy<br/>
</td>
<td>
1 addix<br/>
1 dvossel<br/>
1 jstapleton<br/>
1 manvirr<br/>
1 schmidts<br/>
1 sdolloff<br/>
1 sybasesql<br/>
1 twilson<br/>
1 zerohalo<br/>
</td>
<td>
2 manvirr<br/>
2 RoadKill<br/>
1 addix<br/>
1 anonymouz666<br/>
1 jstapleton<br/>
1 klaus3000<br/>
1 kobaz<br/>
1 nic_bellamy<br/>
1 nickb<br/>
1 sdolloff<br/>
1 sybasesql<br/>
1 wuwu<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/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17080">#17080</a>: [patch] Asterisk crashes while core restart (#0 0x000000000050683c in term_beep (el=0x16cdd9b0) at term.c:865)<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=278981">278981</a><br/>
Reporter: sybasesql<br/>
Testers: sybasesql<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Applications/app_chanspy</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17630">#17630</a>: [patch] Chanspy Keeps using G729 Encoder licenses even after the spying channel hangs up.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=279945">279945</a><br/>
Reporter: manvirr<br/>
Coders: dvossel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17630">#17630</a>: [patch] Chanspy Keeps using G729 Encoder licenses even after the spying channel hangs up.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280448">280448</a><br/>
Reporter: manvirr<br/>
Testers: manvirr, dvossel<br/>
Coders: dvossel<br/>
<br/>
<h3>Category: Applications/app_dial</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17641">#17641</a>: [patch] reset visible_indication after call answering<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281566">281566</a><br/>
Reporter: klaus3000<br/>
Testers: schmidts<br/>
Coders: klaus3000<br/>
<br/>
<h3>Category: Applications/app_disa</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=16661">#16661</a>: [patch] DISA doesn't honor caller ID on the channel<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280811">280811</a><br/>
Reporter: jstapleton<br/>
Testers: jstapleton<br/>
Coders: tilghman<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17874">#17874</a>: [patch] Q931 - Sending PROGRESS after sending ALERTING is a protocol error<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=283048">283048</a><br/>
Reporter: nic_bellamy<br/>
Coders: nic<br/>
<br/>
<h3>Category: Channels/chan_iax2</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17138">#17138</a>: [patch] CallerID not properly set when using Originate and AGI<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281390">281390</a><br/>
Reporter: kobaz<br/>
Coders: jpeeler<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17497">#17497</a>: [patch] [regression] Segmentation fault in scheduled event<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281185">281185</a><br/>
Reporter: anonymouz666<br/>
Coders: dvossel<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17643">#17643</a>: [patch] dialplan reload deadlocks in ast_rdlock_contexts when calling ast_hint_state_changed<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280982">280982</a><br/>
Reporter: zerohalo<br/>
Testers: zerohalo<br/>
Coders: tilghman<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17712">#17712</a>: TOS_SIP does not get set<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=282893">282893</a><br/>
Reporter: nickb<br/>
Coders: dvossel<br/>
<br/>
<h3>Category: Channels/chan_sip/Transfers</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17007">#17007</a>: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=282430">282430</a><br/>
Reporter: addix<br/>
Testers: addix, twilson<br/>
Coders: addix, twilson<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17833">#17833</a>: [patch] say.conf has problem with large numbers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281762">281762</a><br/>
Reporter: RoadKill<br/>
Coders: RoadKill<br/>
<br/>
<a href="https://issues.asterisk.org/view.php?id=17836">#17836</a>: [patch] say.conf added support for Danish<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281819">281819</a><br/>
Reporter: RoadKill<br/>
Coders: RoadKill<br/>
<br/>
<h3>Category: Core/RTP</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17404">#17404</a>: [patch] [regression] audio delay when bridging calls related to timestamp mismatch<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=281911">281911</a><br/>
Reporter: sdolloff<br/>
Testers: sdolloff<br/>
Coders: jpeeler<br/>
<br/>
<h3>Category: General</h3><br/>
<a href="https://issues.asterisk.org/view.php?id=17568">#17568</a>: [patch] DNID does not get cleard on new call<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=278701">278701</a><br/>
Reporter: wuwu<br/>
Coders: rmudgett<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=278984">278984</a></td><td>tilghman</td><td>Establish a maximum version for openh323 (i.e. not opal), because chan_h323 will fail to load, even if it links.</td>
<td><a href="https://issues.asterisk.org/view.php?id=17679">#17679</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=279053">279053</a></td><td>mmichelson</td><td>Backport fixes for sip_uri_params_cmp() from trunk.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=279206">279206</a></td><td>rmudgett</td><td>SIP promiscuous redirect could fail to dial the redirect.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=279344">279344</a></td><td>jpeeler</td><td>Provide a default value for DAHDI_TRANSCODE so when DAHDI is not installed</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=279346">279346</a></td><td>snuffy</td><td>Minor update to man page</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280088">280088</a></td><td>lmadsen</td><td>Update help text to be less confusing.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280341">280341</a></td><td>jeang</td><td>Fix a dsp structure leak occuring when a local channel is put into a meetme</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=280944">280944</a></td><td>russell</td><td>Copy astcli back to 1.4 so it's available for automated testing purposes.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=282129">282129</a></td><td>qwell</td><td>Register CLI commands before parsing config, in case there is a config error.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=282729">282729</a></td><td>twilson</td><td>Add some documentation about codec negotiation to sip.conf</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/1.4?view=revision&revision=283123">283123</a></td><td>rmudgett</td><td>Merged revision 278274 from</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_dial.c | 12 +-
apps/app_meetme.c | 3
apps/app_queue.c | 9 +
autoconf/ast_check_pwlib.m4 | 12 ++
channels/chan_dahdi.c | 18 +--
channels/chan_local.c | 20 +++
channels/chan_sip.c | 204 +++++++++++++++++++++++++++------------
configs/say.conf.sample | 96 +++++++++++++++++-
configs/sip.conf.sample | 13 ++
configure.ac | 3
contrib/scripts/astcli | 167 +++++++++++++++++++++++++++++++
contrib/scripts/live_ast | 2
doc/asterisk.8 | 4
funcs/func_callerid.c | 6 +
include/asterisk/audiohook.h | 7 +
include/asterisk/autoconfig.h.in | 51 ++++-----
main/asterisk.c | 19 ++-
main/audiohook.c | 12 ++
main/channel.c | 27 ++++-
main/pbx.c | 14 +-
pbx/pbx_config.c | 7 -
21 files changed, 572 insertions(+), 134 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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Release Summary
asterisk-1.4.36
Date: 2010-09-13
<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-1.4.35.
----------------------------------------------------------------------
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
4 dvossel 1 addix 2 manvirr
4 tilghman 1 dvossel 2 RoadKill
3 jpeeler 1 jstapleton 1 addix
3 rmudgett 1 manvirr 1 anonymouz666
2 RoadKill 1 schmidts 1 jstapleton
2 twilson 1 sdolloff 1 klaus3000
1 addix 1 sybasesql 1 kobaz
1 jeang 1 twilson 1 nic_bellamy
1 klaus3000 1 zerohalo 1 nickb
1 lmadsen 1 sdolloff
1 mmichelson 1 sybasesql
1 nic 1 wuwu
1 qwell 1 zerohalo
1 russell
1 snuffy
----------------------------------------------------------------------
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/General
#17080: [patch] Asterisk crashes while core restart (#0 0x000000000050683c
in term_beep (el=0x16cdd9b0) at term.c:865)
Revision: 278981
Reporter: sybasesql
Testers: sybasesql
Coders: tilghman
Category: Applications/app_chanspy
#17630: [patch] Chanspy Keeps using G729 Encoder licenses even after the
spying channel hangs up.
Revision: 279945
Reporter: manvirr
Coders: dvossel
#17630: [patch] Chanspy Keeps using G729 Encoder licenses even after the
spying channel hangs up.
Revision: 280448
Reporter: manvirr
Testers: manvirr, dvossel
Coders: dvossel
Category: Applications/app_dial
#17641: [patch] reset visible_indication after call answering
Revision: 281566
Reporter: klaus3000
Testers: schmidts
Coders: klaus3000
Category: Applications/app_disa
#16661: [patch] DISA doesn't honor caller ID on the channel
Revision: 280811
Reporter: jstapleton
Testers: jstapleton
Coders: tilghman
Category: Channels/chan_dahdi
#17874: [patch] Q931 - Sending PROGRESS after sending ALERTING is a
protocol error
Revision: 283048
Reporter: nic_bellamy
Coders: nic
Category: Channels/chan_iax2
#17138: [patch] CallerID not properly set when using Originate and AGI
Revision: 281390
Reporter: kobaz
Coders: jpeeler
Category: Channels/chan_sip/General
#17497: [patch] [regression] Segmentation fault in scheduled event
Revision: 281185
Reporter: anonymouz666
Coders: dvossel
#17643: [patch] dialplan reload deadlocks in ast_rdlock_contexts when
calling ast_hint_state_changed
Revision: 280982
Reporter: zerohalo
Testers: zerohalo
Coders: tilghman
#17712: TOS_SIP does not get set
Revision: 282893
Reporter: nickb
Coders: dvossel
Category: Channels/chan_sip/Transfers
#17007: [patch] RTP Timestamp changes after transfer, but SSRC not and the
markerbit ist not set.
Revision: 282430
Reporter: addix
Testers: addix, twilson
Coders: addix, twilson
Category: Core/Configuration
#17833: [patch] say.conf has problem with large numbers
Revision: 281762
Reporter: RoadKill
Coders: RoadKill
#17836: [patch] say.conf added support for Danish
Revision: 281819
Reporter: RoadKill
Coders: RoadKill
Category: Core/RTP
#17404: [patch] [regression] audio delay when bridging calls related to
timestamp mismatch
Revision: 281911
Reporter: sdolloff
Testers: sdolloff
Coders: jpeeler
Category: General
#17568: [patch] DNID does not get cleard on new call
Revision: 278701
Reporter: wuwu
Coders: rmudgett
----------------------------------------------------------------------
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 |
|----------+------------+-----------------------------------+------------|
| | | Establish a maximum version for | |
| 278984 | tilghman | openh323 (i.e. not opal), because | #17679 |
| | | chan_h323 will fail to load, even | |
| | | if it links. | |
|----------+------------+-----------------------------------+------------|
| 279053 | mmichelson | Backport fixes for | |
| | | sip_uri_params_cmp() from trunk. | |
|----------+------------+-----------------------------------+------------|
| 279206 | rmudgett | SIP promiscuous redirect could | |
| | | fail to dial the redirect. | |
|----------+------------+-----------------------------------+------------|
| | | Provide a default value for | |
| 279344 | jpeeler | DAHDI_TRANSCODE so when DAHDI is | |
| | | not installed | |
|----------+------------+-----------------------------------+------------|
| 279346 | snuffy | Minor update to man page | |
|----------+------------+-----------------------------------+------------|
| 280088 | lmadsen | Update help text to be less | |
| | | confusing. | |
|----------+------------+-----------------------------------+------------|
| | | Fix a dsp structure leak occuring | |
| 280341 | jeang | when a local channel is put into | |
| | | a meetme | |
|----------+------------+-----------------------------------+------------|
| | | Copy astcli back to 1.4 so it's | |
| 280944 | russell | available for automated testing | |
| | | purposes. | |
|----------+------------+-----------------------------------+------------|
| | | Register CLI commands before | |
| 282129 | qwell | parsing config, in case there is | |
| | | a config error. | |
|----------+------------+-----------------------------------+------------|
| 282729 | twilson | Add some documentation about | |
| | | codec negotiation to sip.conf | |
|----------+------------+-----------------------------------+------------|
| 283123 | rmudgett | Merged revision 278274 from | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
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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_dial.c | 12 +-
apps/app_meetme.c | 3
apps/app_queue.c | 9 +
autoconf/ast_check_pwlib.m4 | 12 ++
channels/chan_dahdi.c | 18 +--
channels/chan_local.c | 20 +++
channels/chan_sip.c | 204 +++++++++++++++++++++++++++------------
configs/say.conf.sample | 96 +++++++++++++++++-
configs/sip.conf.sample | 13 ++
configure.ac | 3
contrib/scripts/astcli | 167 +++++++++++++++++++++++++++++++
contrib/scripts/live_ast | 2
doc/asterisk.8 | 4
funcs/func_callerid.c | 6 +
include/asterisk/audiohook.h | 7 +
include/asterisk/autoconfig.h.in | 51 ++++-----
main/asterisk.c | 19 ++-
main/audiohook.c | 12 ++
main/channel.c | 27 ++++-
main/pbx.c | 14 +-
pbx/pbx_config.c | 7 -
21 files changed, 572 insertions(+), 134 deletions(-)
----------------------------------------------------------------------