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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1@369935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Importing release summary for 10.7.0-digiumphones-rc1 release.
2026-01-10 03:31:06 +00:00 · 2012-07-11 15:53:48 +00:00 · 2012-07-11 15:53:37 +00:00 · 2012-07-11 15:53:29 +00:00 · 2012-07-11 15:52:03 +00:00
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+<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-10.7.0-digiumphones-rc1</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">asterisk-10.7.0-digiumphones-rc1</h3>
+<h3 align="center">Date: 2012-07-11</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-10.6.0-digiumphones.</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>
+29 root<br/>
+9 qwell<br/>
+4 Mark<br/>
+1 file<br/>
+1 Michael<br/>
+1 rmudgett<br/>
+</td>
+<td>
+2 Guenther Kelleter<br/>
+1 jamicque<br/>
+1 Michael L. Young<br/>
+1 Paul Belanger<br/>
+1 rmudgett<br/>
+1 Steve Davies<br/>
+1 Terry Wilson<br/>
+1 Tilghman Lesher<br/>
+</td>
+<td>
+3 lmadsen<br/>
+2 fnordian<br/>
+2 one47<br/>
+1 alecdavis<br/>
+1 drdelaney<br/>
+1 elguero<br/>
+1 jamicque<br/>
+1 karlfife<br/>
+1 mdavenport<br/>
+1 mjordan<br/>
+1 mmichelson<br/>
+1 sdolloff<br/>
+1 themsley<br/>
+1 tomaso<br/>
+1 tsarik<br/>
+1 vsauer<br/>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="issues"><h2 align="center">Closed Issues</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
+<h3>Category: Addons/chan_ooh323</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369107">369107</a><br/>
+Reporter: tsarik<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Applications/app_dial</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19611">ASTERISK-19611</a>: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368917">368917</a><br/>
+Reporter: vsauer<br/>
+Coders: Mark<br/>
+<br/>
+<h3>Category: Applications/app_voicemail</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19923">ASTERISK-19923</a>: Asterisk crashing due to memory corruptions in chan_sip/voicemail<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369665">369665</a><br/>
+Reporter: drdelaney<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Channels/chan_iax2</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19801">ASTERISK-19801</a>: Deadlock with masquerade and chan_iax<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368823">368823</a><br/>
+Reporter: alecdavis<br/>
+Testers: Guenther Kelleter<br/>
+Coders: qwell<br/>
+<br/>
+<h3>Category: Channels/chan_local</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19611">ASTERISK-19611</a>: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368917">368917</a><br/>
+Reporter: vsauer<br/>
+Coders: Mark<br/>
+<br/>
+<h3>Category: Channels/chan_sip/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19179">ASTERISK-19179</a>: RTP inactivity SIP / ooh323 wont work<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369107">369107</a><br/>
+Reporter: tsarik<br/>
+Coders: root<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19859">ASTERISK-19859</a>: cid_tag is not set according to the sip configuration anymore if get_rpid() != 0<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368823">368823</a><br/>
+Reporter: tomaso<br/>
+Testers: Guenther Kelleter<br/>
+Coders: qwell<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19892">ASTERISK-19892</a>: If Asterisk sends a 481 to an initial INVITE that contained a to-tag, then Asterisk will not recognize the ensuing ACK<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369364">369364</a><br/>
+Reporter: mmichelson<br/>
+Coders: root<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369453">369453</a><br/>
+Reporter: one47<br/>
+Coders: root<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a>: SIP re-INVITEs have no transaction timeout<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369578">369578</a><br/>
+Reporter: one47<br/>
+Testers: Steve Davies, Terry Wilson<br/>
+Coders: root<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20008">ASTERISK-20008</a>: outboundproxy ignored after when sending invite after 407<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369083">369083</a><br/>
+Reporter: fnordian<br/>
+Coders: Mark<br/>
+<br/>
+<h3>Category: Channels/chan_sip/IPv6</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16618">ASTERISK-16618</a>: Unable to use IPv4 addresses for a TCP host when using IPv6<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369488">369488</a><br/>
+Reporter: lmadsen<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19601">ASTERISK-19601</a>: Failure of domain matching on authentication of INVITE request produces misleading NOTICE message; bypasses alwaysauthreject logic<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369322">369322</a><br/>
+Reporter: mjordan<br/>
+Coders: Mark<br/>
+<br/>
+<h3>Category: Core/Configuration</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19910">ASTERISK-19910</a>: Add sip_notify.conf entry for Digium phones<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369836">369836</a><br/>
+Reporter: mdavenport<br/>
+Coders: root<br/>
+<br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19920">ASTERISK-19920</a>: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368893">368893</a><br/>
+Reporter: lmadsen<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Core/General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19834">ASTERISK-19834</a>: Memory leak caused by thread created by bridge_channel_join being neither joined nor detached<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369725">369725</a><br/>
+Reporter: fnordian<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Core/Netsock</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20006">ASTERISK-20006</a>: Fix NULL pointer segfault in ast_sockaddr_parse()<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369125">369125</a><br/>
+Reporter: elguero<br/>
+Testers: Michael L. Young<br/>
+Coders: Michael<br/>
+<br/>
+<h3>Category: Documentation</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20007">ASTERISK-20007</a>: GotoIf() documentation updates to be more clear that [[context,]extension,]priority is valid<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369889">369889</a><br/>
+Reporter: lmadsen<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: General</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19492">ASTERISK-19492</a>: Group write permission removed from existing directory /etc/asterisk/. when updating <br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368872">368872</a><br/>
+Reporter: karlfife<br/>
+Testers: Paul Belanger, Tilghman Lesher<br/>
+Coders: root<br/>
+<br/>
+<h3>Category: Resources/res_adsi</h3><br/>
+<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19920">ASTERISK-19920</a>: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions<br/>
+Revision: <a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368893">368893</a><br/>
+Reporter: lmadsen<br/>
+Coders: root<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/10-digiumphones?view=revision&revision=368810">368810</a></td><td>qwell</td><td>enable automerge</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368811">368811</a></td><td>qwell</td><td>Let's try using an automerge-propname, since we have multiple heads.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368826">368826</a></td><td>qwell</td><td>Let's fix the 1.8-merged prop, to give automerge the best chance at succeeding.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368946">368946</a></td><td>root</td><td>Revert Makefile change to remove embedding res_adsi.so</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368960">368960</a></td><td>root</td><td>AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19905">ASTERISK-19905</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368963">368963</a></td><td>qwell</td><td>Remove global symbol requirement from app_voicemail.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368965">368965</a></td><td>qwell</td><td>These functions that were moved need to be static.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=368999">368999</a></td><td>qwell</td><td>Remove some symbol exports that got missed in the removal of global symbols.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369023">369023</a></td><td>root</td><td>Multiple revisions 369001-369002</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369026">369026</a></td><td>qwell</td><td>Fix voicemail API tests by using the correct argument order for create/destroy.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369164">369164</a></td><td>root</td><td>fix locking issue on empty callList</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19298">ASTERISK-19298</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369213">369213</a></td><td>root</td><td>Don't parse media stream state for SIP video streams</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369234">369234</a></td><td>root</td><td>Don't crash on a guest directmedia call</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369257">369257</a></td><td>root</td><td>Change incorrect chan_sip zombie hangup debug message.  They are all zombies now.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369280">369280</a></td><td>root</td><td>Check if PBX was started and fix F and F(x) action logic in Dial application.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369294">369294</a></td><td>root</td><td>Fix Bridge application and AMI Bridge action error handling.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369384">369384</a></td><td>root</td><td>Fix incorrect duration reporting in CDRs created in batch mode</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19860">ASTERISK-19860</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369406">369406</a></td><td>root</td><td>Fix crash in unloading of res_adsi module</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369508">369508</a></td><td>root</td><td>With some configurations a transport is not actually specified so assume UDP in these cases.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369516">369516</a></td><td>root</td><td>Fix apparent copy and paste error where incorrect "glue" is used.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369598">369598</a></td><td>root</td><td>More improvements to re-INVITEs timing out after a provisional response</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369643">369643</a></td><td>root</td><td>Do not send a BYE when a provisional response arrives during a re-INVITE</td>
+<td><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19992">ASTERISK-19992</a></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369763">369763</a></td><td>root</td><td>chan_sip: Add case for FLASH control frames so that we don't display a warning.</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369810">369810</a></td><td>root</td><td>chan_sip: Fix small behavioral change accidentally introduced in r369750</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/10-digiumphones?view=revision&revision=369846">369846</a></td><td>file</td><td>Add support for exposing the received contact URI and also for setting the request URI in messages.</td>
+<td></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+Makefile                               |   46 +--
+addons/chan_ooh323.c                   |   23 +
+addons/ooh323c/src/ooCalls.c           |    3
+addons/ooh323c/src/ooq931.c            |    2
+apps/app_dial.c                        |   34 +-
+apps/app_directory.c                   |    3
+apps/app_queue.c                       |   14 -
+apps/app_stack.c                       |    5
+apps/app_voicemail.c                   |  201 +++++++++-------
+apps/app_voicemail.exports.in          |    9
+apps/confbridge/conf_config_parser.c   |    4
+build_tools/find_missing_support_level |    3
+channels/chan_dahdi.c                  |   16 -
+channels/chan_iax2.c                   |   15 -
+channels/chan_misdn.c                  |    1
+channels/chan_sip.c                    |  251 ++++++++++++++------
+channels/chan_skinny.c                 |   14 -
+channels/console_board.c               |    4
+channels/console_gui.c                 |    4
+channels/console_video.c               |    4
+channels/iax2-parser.c                 |    4
+channels/iax2-provision.c              |    4
+channels/misdn/ie.c                    |    4
+channels/misdn/isdn_lib.c              |    4
+channels/misdn/isdn_msg_parser.c       |    4
+channels/misdn/portinfo.c              |    3
+channels/misdn_config.c                |    4
+channels/sig_analog.c                  |   15 +
+channels/sig_pri.c                     |    3
+channels/sig_ss7.c                     |    3
+channels/sip/config_parser.c           |    4
+channels/sip/dialplan_functions.c      |    8
+channels/sip/include/sip.h             |    4
+channels/sip/reqresp_parser.c          |    6
+channels/sip/sdp_crypto.c              |    8
+channels/sip/security_events.c         |    4
+channels/sip/srtp.c                    |    4
+channels/vcodecs.c                     |    4
+channels/vgrabbers.c                   |    4
+configs/sip_notify.conf.sample         |    5
+funcs/func_strings.c                   |    3
+funcs/func_volume.c                    |    3
+include/asterisk/adsi.h                |   93 +++++--
+include/asterisk/app.h                 |  215 +++++++++++++++++
+include/asterisk/app_voicemail.h       |  213 -----------------
+include/asterisk/channel.h             |    2
+include/asterisk/netsock2.h            |    3
+main/Makefile                          |    3
+main/abstract_jb.c                     |    4
+main/acl.c                             |    4
+main/adsi.c                            |  351 ++++++++++++++++++++++++++++
+main/alaw.c                            |    4
+main/aoc.c                             |    4
+main/app.c                             |  210 ++++++++++++++++-
+main/asterisk.c                        |    4
+main/astfd.c                           |    4
+main/astmm.c                           |    4
+main/astobj2.c                         |    5
+main/audiohook.c                       |    4
+main/autochan.c                        |    4
+main/autoservice.c                     |    4
+main/bridging.c                        |   18 -
+main/callerid.c                        |    4
+main/ccss.c                            |   13 -
+main/cdr.c                             |   10
+main/cel.c                             |    4
+main/channel.c                         |   14 -
+main/chanvars.c                        |    4
+main/cli.c                             |    4
+main/config.c                          |    4
+main/data.c                            |    4
+main/datastore.c                       |    4
+main/db.c                              |    4
+main/devicestate.c                     |    4
+main/dial.c                            |    4
+main/dns.c                             |    4
+main/dnsmgr.c                          |    4
+main/dsp.c                             |    4
+main/enum.c                            |    4
+main/event.c                           |    4
+main/features.c                        |  407 ++++++++++++++++++---------------
+main/file.c                            |    4
+main/fixedjitterbuf.c                  |    4
+main/format.c                          |    4
+main/format_cap.c                      |    4
+main/format_pref.c                     |    4
+main/frame.c                           |    4
+main/framehook.c                       |    4
+main/fskmodem.c                        |    4
+main/fskmodem_float.c                  |    4
+main/fskmodem_int.c                    |    4
+main/global_datastores.c               |    4
+main/hashtab.c                         |    4
+main/heap.c                            |    4
+main/image.c                           |    4
+main/indications.c                     |    4
+main/io.c                              |    4
+main/jitterbuf.c                       |    4
+main/loader.c                          |    8
+main/lock.c                            |    4
+main/logger.c                          |    4
+main/md5.c                             |    6
+main/message.c                         |    4
+main/netsock.c                         |    4
+main/netsock2.c                        |   10
+main/pbx.c                             |   24 +
+main/plc.c                             |    4
+main/privacy.c                         |    4
+main/rtp_engine.c                      |    6
+main/say.c                             |    6
+main/sched.c                           |    4
+main/security_events.c                 |    4
+main/slinfactory.c                     |    4
+main/srv.c                             |    4
+main/ssl.c                             |    4
+main/stdtime/localtime.c               |    4
+main/strcompat.c                       |    4
+main/strings.c                         |    4
+main/stun.c                            |    4
+main/syslog.c                          |    4
+main/taskprocessor.c                   |    4
+main/tcptls.c                          |    7
+main/tdd.c                             |    4
+main/term.c                            |    4
+main/test.c                            |    4
+main/threadstorage.c                   |    4
+main/timing.c                          |    4
+main/translate.c                       |    4
+main/udptl.c                           |    7
+main/ulaw.c                            |    4
+main/utils.c                           |    4
+main/xml.c                             |    4
+main/xmldoc.c                          |    4
+pbx/dundi-parser.c                     |    4
+pbx/pbx_config.c                       |    4
+res/ael/pval.c                         |    4
+res/ais/clm.c                          |    4
+res/ais/evt.c                          |    4
+res/res_adsi.c                         |  187 ++++++++++-----
+res/res_adsi.exports.in                |   33 --
+res/res_config_odbc.c                  |    7
+res/res_fax.c                          |    2
+res/res_odbc.c                         |    2
+res/res_smdi.c                         |    2
+res/res_speech.c                       |    3
+res/snmp/agent.c                       |    4
+tests/test_voicemail_api.c             |    1
+utils/astdb2bdb.c                      |    6
+utils/astdb2sqlite3.c                  |    6
+149 files changed, 2131 insertions(+), 810 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>
--- a/asterisk-10.7.0-digiumphones-rc1-summary.txt
+++ b/asterisk-10.7.0-digiumphones-rc1-summary.txt
@@ -0,0 +1,491 @@
+                                Release Summary
+
+                        asterisk-10.7.0-digiumphones-rc1
+
+                                Date: 2012-07-11
+
+                           <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-10.6.0-digiumphones.
+
+     ----------------------------------------------------------------------
+
+                                  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              
+   29 root                  2 Guenther Kelleter      3 lmadsen                
+   9 qwell                  1 jamicque               2 fnordian               
+   4 Mark                   1 Michael L. Young       2 one47                  
+   1 file                   1 Paul Belanger          1 alecdavis              
+   1 Michael                1 rmudgett               1 drdelaney              
+   1 rmudgett               1 Steve Davies           1 elguero                
+                            1 Terry Wilson           1 jamicque               
+                            1 Tilghman Lesher        1 karlfife               
+                                                     1 mdavenport             
+                                                     1 mjordan                
+                                                     1 mmichelson             
+                                                     1 sdolloff               
+                                                     1 themsley               
+                                                     1 tomaso                 
+                                                     1 tsarik                 
+                                                     1 vsauer                 
+
+     ----------------------------------------------------------------------
+
+                                 Closed Issues
+
+                                 [Back to Top]
+
+   This is a list of all issues from the issue tracker that were closed by
+   changes that went into this release.
+
+  Category: Addons/chan_ooh323
+
+   ASTERISK-19179: RTP inactivity SIP / ooh323 wont work
+   Revision: 369107
+   Reporter: tsarik
+   Coders: root
+
+  Category: Applications/app_dial
+
+   ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom
+   phone is redirected by "302 Moved temporarily" to chan_local
+   Revision: 368917
+   Reporter: vsauer
+   Coders: Mark
+
+  Category: Applications/app_voicemail
+
+   ASTERISK-19923: Asterisk crashing due to memory corruptions in
+   chan_sip/voicemail
+   Revision: 369665
+   Reporter: drdelaney
+   Coders: root
+
+  Category: Channels/chan_iax2
+
+   ASTERISK-19801: Deadlock with masquerade and chan_iax
+   Revision: 368823
+   Reporter: alecdavis
+   Testers: Guenther Kelleter
+   Coders: qwell
+
+  Category: Channels/chan_local
+
+   ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom
+   phone is redirected by "302 Moved temporarily" to chan_local
+   Revision: 368917
+   Reporter: vsauer
+   Coders: Mark
+
+  Category: Channels/chan_sip/General
+
+   ASTERISK-19179: RTP inactivity SIP / ooh323 wont work
+   Revision: 369107
+   Reporter: tsarik
+   Coders: root
+
+   ASTERISK-19859: cid_tag is not set according to the sip configuration
+   anymore if get_rpid() != 0
+   Revision: 368823
+   Reporter: tomaso
+   Testers: Guenther Kelleter
+   Coders: qwell
+
+   ASTERISK-19892: If Asterisk sends a 481 to an initial INVITE that
+   contained a to-tag, then Asterisk will not recognize the ensuing ACK
+   Revision: 369364
+   Reporter: mmichelson
+   Coders: root
+
+   ASTERISK-19992: SIP re-INVITEs have no transaction timeout
+   Revision: 369453
+   Reporter: one47
+   Coders: root
+
+   ASTERISK-19992: SIP re-INVITEs have no transaction timeout
+   Revision: 369578
+   Reporter: one47
+   Testers: Steve Davies, Terry Wilson
+   Coders: root
+
+   ASTERISK-20008: outboundproxy ignored after when sending invite after 407
+   Revision: 369083
+   Reporter: fnordian
+   Coders: Mark
+
+  Category: Channels/chan_sip/IPv6
+
+   ASTERISK-16618: Unable to use IPv4 addresses for a TCP host when using
+   IPv6
+   Revision: 369488
+   Reporter: lmadsen
+   Coders: root
+
+  Category: Channels/chan_sip/Interoperability
+
+   ASTERISK-19601: Failure of domain matching on authentication of INVITE
+   request produces misleading NOTICE message; bypasses alwaysauthreject
+   logic
+   Revision: 369322
+   Reporter: mjordan
+   Coders: Mark
+
+  Category: Core/Configuration
+
+   ASTERISK-19910: Add sip_notify.conf entry for Digium phones
+   Revision: 369836
+   Reporter: mdavenport
+   Coders: root
+
+   ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is)
+   despite no modules.conf, noload or autoload=no instructions
+   Revision: 368893
+   Reporter: lmadsen
+   Coders: root
+
+  Category: Core/General
+
+   ASTERISK-19834: Memory leak caused by thread created by
+   bridge_channel_join being neither joined nor detached
+   Revision: 369725
+   Reporter: fnordian
+   Coders: root
+
+  Category: Core/Netsock
+
+   ASTERISK-20006: Fix NULL pointer segfault in ast_sockaddr_parse()
+   Revision: 369125
+   Reporter: elguero
+   Testers: Michael L. Young
+   Coders: Michael
+
+  Category: Documentation
+
+   ASTERISK-20007: GotoIf() documentation updates to be more clear that
+   [[context,]extension,]priority is valid
+   Revision: 369889
+   Reporter: lmadsen
+   Coders: root
+
+  Category: General
+
+   ASTERISK-19492: Group write permission removed from existing directory
+   /etc/asterisk/. when updating
+   Revision: 368872
+   Reporter: karlfife
+   Testers: Paul Belanger, Tilghman Lesher
+   Coders: root
+
+  Category: Resources/res_adsi
+
+   ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is)
+   despite no modules.conf, noload or autoload=no instructions
+   Revision: 368893
+   Reporter: lmadsen
+   Coders: root
+
+     ----------------------------------------------------------------------
+
+                      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 |
+   |----------+--------+--------------------------------+-------------------|
+   | 368810   | qwell  | enable automerge               |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Let's try using an             |                   |
+   | 368811   | qwell  | automerge-propname, since we   |                   |
+   |          |        | have multiple heads.           |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Let's fix the 1.8-merged prop, |                   |
+   | 368826   | qwell  | to give automerge the best     |                   |
+   |          |        | chance at succeeding.          |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 368946   | root   | Revert Makefile change to      |                   |
+   |          |        | remove embedding res_adsi.so   |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | AST-2012-009: Fix crash in     |                   |
+   | 368960   | root   | chan_skinny due to Key Pad     | ASTERISK-19905    |
+   |          |        | Button Message handling        |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Remove global symbol           |                   |
+   | 368963   | qwell  | requirement from               |                   |
+   |          |        | app_voicemail.                 |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 368965   | qwell  | These functions that were      |                   |
+   |          |        | moved need to be static.       |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Remove some symbol exports     |                   |
+   | 368999   | qwell  | that got missed in the removal |                   |
+   |          |        | of global symbols.             |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369023   | root   | Multiple revisions             |                   |
+   |          |        | 369001-369002                  |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Fix voicemail API tests by     |                   |
+   | 369026   | qwell  | using the correct argument     |                   |
+   |          |        | order for create/destroy.      |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369164   | root   | fix locking issue on empty     | ASTERISK-19298    |
+   |          |        | callList                       |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369213   | root   | Don't parse media stream state |                   |
+   |          |        | for SIP video streams          |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369234   | root   | Don't crash on a guest         |                   |
+   |          |        | directmedia call               |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Change incorrect chan_sip      |                   |
+   | 369257   | root   | zombie hangup debug message.   |                   |
+   |          |        | They are all zombies now.      |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Check if PBX was started and   |                   |
+   | 369280   | root   | fix F and F(x) action logic in |                   |
+   |          |        | Dial application.              |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369294   | root   | Fix Bridge application and AMI |                   |
+   |          |        | Bridge action error handling.  |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Fix incorrect duration         |                   |
+   | 369384   | root   | reporting in CDRs created in   | ASTERISK-19860    |
+   |          |        | batch mode                     |                   |
+   |----------+--------+--------------------------------+-------------------|
+   | 369406   | root   | Fix crash in unloading of      |                   |
+   |          |        | res_adsi module                |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | With some configurations a     |                   |
+   | 369508   | root   | transport is not actually      |                   |
+   |          |        | specified so assume UDP in     |                   |
+   |          |        | these cases.                   |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Fix apparent copy and paste    |                   |
+   | 369516   | root   | error where incorrect "glue"   |                   |
+   |          |        | is used.                       |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | More improvements to           |                   |
+   | 369598   | root   | re-INVITEs timing out after a  | ASTERISK-19992    |
+   |          |        | provisional response           |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Do not send a BYE when a       |                   |
+   | 369643   | root   | provisional response arrives   | ASTERISK-19992    |
+   |          |        | during a re-INVITE             |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | chan_sip: Add case for FLASH   |                   |
+   | 369763   | root   | control frames so that we      |                   |
+   |          |        | don't display a warning.       |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | chan_sip: Fix small behavioral |                   |
+   | 369810   | root   | change accidentally introduced |                   |
+   |          |        | in r369750                     |                   |
+   |----------+--------+--------------------------------+-------------------|
+   |          |        | Add support for exposing the   |                   |
+   | 369846   | file   | received contact URI and also  |                   |
+   |          |        | for setting the request URI in |                   |
+   |          |        | messages.                      |                   |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ Makefile                               |   46 +--
+ addons/chan_ooh323.c                   |   23 +
+ addons/ooh323c/src/ooCalls.c           |    3
+ addons/ooh323c/src/ooq931.c            |    2
+ apps/app_dial.c                        |   34 +-
+ apps/app_directory.c                   |    3
+ apps/app_queue.c                       |   14 -
+ apps/app_stack.c                       |    5
+ apps/app_voicemail.c                   |  201 +++++++++-------
+ apps/app_voicemail.exports.in          |    9
+ apps/confbridge/conf_config_parser.c   |    4
+ build_tools/find_missing_support_level |    3
+ channels/chan_dahdi.c                  |   16 -
+ channels/chan_iax2.c                   |   15 -
+ channels/chan_misdn.c                  |    1
+ channels/chan_sip.c                    |  251 ++++++++++++++------
+ channels/chan_skinny.c                 |   14 -
+ channels/console_board.c               |    4
+ channels/console_gui.c                 |    4
+ channels/console_video.c               |    4
+ channels/iax2-parser.c                 |    4
+ channels/iax2-provision.c              |    4
+ channels/misdn/ie.c                    |    4
+ channels/misdn/isdn_lib.c              |    4
+ channels/misdn/isdn_msg_parser.c       |    4
+ channels/misdn/portinfo.c              |    3
+ channels/misdn_config.c                |    4
+ channels/sig_analog.c                  |   15 +
+ channels/sig_pri.c                     |    3
+ channels/sig_ss7.c                     |    3
+ channels/sip/config_parser.c           |    4
+ channels/sip/dialplan_functions.c      |    8
+ channels/sip/include/sip.h             |    4
+ channels/sip/reqresp_parser.c          |    6
+ channels/sip/sdp_crypto.c              |    8
+ channels/sip/security_events.c         |    4
+ channels/sip/srtp.c                    |    4
+ channels/vcodecs.c                     |    4
+ channels/vgrabbers.c                   |    4
+ configs/sip_notify.conf.sample         |    5
+ funcs/func_strings.c                   |    3
+ funcs/func_volume.c                    |    3
+ include/asterisk/adsi.h                |   93 +++++--
+ include/asterisk/app.h                 |  215 +++++++++++++++++
+ include/asterisk/app_voicemail.h       |  213 -----------------
+ include/asterisk/channel.h             |    2
+ include/asterisk/netsock2.h            |    3
+ main/Makefile                          |    3
+ main/abstract_jb.c                     |    4
+ main/acl.c                             |    4
+ main/adsi.c                            |  351 ++++++++++++++++++++++++++++
+ main/alaw.c                            |    4
+ main/aoc.c                             |    4
+ main/app.c                             |  210 ++++++++++++++++-
+ main/asterisk.c                        |    4
+ main/astfd.c                           |    4
+ main/astmm.c                           |    4
+ main/astobj2.c                         |    5
+ main/audiohook.c                       |    4
+ main/autochan.c                        |    4
+ main/autoservice.c                     |    4
+ main/bridging.c                        |   18 -
+ main/callerid.c                        |    4
+ main/ccss.c                            |   13 -
+ main/cdr.c                             |   10
+ main/cel.c                             |    4
+ main/channel.c                         |   14 -
+ main/chanvars.c                        |    4
+ main/cli.c                             |    4
+ main/config.c                          |    4
+ main/data.c                            |    4
+ main/datastore.c                       |    4
+ main/db.c                              |    4
+ main/devicestate.c                     |    4
+ main/dial.c                            |    4
+ main/dns.c                             |    4
+ main/dnsmgr.c                          |    4
+ main/dsp.c                             |    4
+ main/enum.c                            |    4
+ main/event.c                           |    4
+ main/features.c                        |  407 ++++++++++++++++++---------------
+ main/file.c                            |    4
+ main/fixedjitterbuf.c                  |    4
+ main/format.c                          |    4
+ main/format_cap.c                      |    4
+ main/format_pref.c                     |    4
+ main/frame.c                           |    4
+ main/framehook.c                       |    4
+ main/fskmodem.c                        |    4
+ main/fskmodem_float.c                  |    4
+ main/fskmodem_int.c                    |    4
+ main/global_datastores.c               |    4
+ main/hashtab.c                         |    4
+ main/heap.c                            |    4
+ main/image.c                           |    4
+ main/indications.c                     |    4
+ main/io.c                              |    4
+ main/jitterbuf.c                       |    4
+ main/loader.c                          |    8
+ main/lock.c                            |    4
+ main/logger.c                          |    4
+ main/md5.c                             |    6
+ main/message.c                         |    4
+ main/netsock.c                         |    4
+ main/netsock2.c                        |   10
+ main/pbx.c                             |   24 +
+ main/plc.c                             |    4
+ main/privacy.c                         |    4
+ main/rtp_engine.c                      |    6
+ main/say.c                             |    6
+ main/sched.c                           |    4
+ main/security_events.c                 |    4
+ main/slinfactory.c                     |    4
+ main/srv.c                             |    4
+ main/ssl.c                             |    4
+ main/stdtime/localtime.c               |    4
+ main/strcompat.c                       |    4
+ main/strings.c                         |    4
+ main/stun.c                            |    4
+ main/syslog.c                          |    4
+ main/taskprocessor.c                   |    4
+ main/tcptls.c                          |    7
+ main/tdd.c                             |    4
+ main/term.c                            |    4
+ main/test.c                            |    4
+ main/threadstorage.c                   |    4
+ main/timing.c                          |    4
+ main/translate.c                       |    4
+ main/udptl.c                           |    7
+ main/ulaw.c                            |    4
+ main/utils.c                           |    4
+ main/xml.c                             |    4
+ main/xmldoc.c                          |    4
+ pbx/dundi-parser.c                     |    4
+ pbx/pbx_config.c                       |    4
+ res/ael/pval.c                         |    4
+ res/ais/clm.c                          |    4
+ res/ais/evt.c                          |    4
+ res/res_adsi.c                         |  187 ++++++++++-----
+ res/res_adsi.exports.in                |   33 --
+ res/res_config_odbc.c                  |    7
+ res/res_fax.c                          |    2
+ res/res_odbc.c                         |    2
+ res/res_smdi.c                         |    2
+ res/res_speech.c                       |    3
+ res/snmp/agent.c                       |    4
+ tests/test_voicemail_api.c             |    1
+ utils/astdb2bdb.c                      |    6
+ utils/astdb2sqlite3.c                  |    6
+ 149 files changed, 2131 insertions(+), 810 deletions(-)
+
+     ----------------------------------------------------------------------
Author	SHA1	Message	Date
Asterisk Autobuilder	c866728479	Use autotagged externals git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1@369935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-07-11 15:53:48 +00:00
Asterisk Autobuilder	ebc271cb46	Importing release summary for 10.7.0-digiumphones-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1@369934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-07-11 15:53:37 +00:00
Asterisk Autobuilder	5a128db9c3	Importing files for 10.7.0-digiumphones-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1@369933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-07-11 15:53:29 +00:00
Asterisk Autobuilder	db4044236f	Creating tag for the release of asterisk-10.7.0-digiumphones-rc1 git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1@369932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2012-07-11 15:52:03 +00:00