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Asterisk Development Team
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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"><title>Release Summary - asterisk-13.32.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.32.0-rc1</h3><h3 align="center">Date: 2020-03-05</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="#closed_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><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version 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-13.31.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><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 affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">10 Sean Bright <sean.bright@gmail.com><br/>5 Walter Doekes <walter+asterisk@wjd.nu><br/>4 Kevin Harwell <kharwell@digium.com><br/>2 Torrey Searle <torrey@voxbone.com><br/>2 Joshua C. Colp <jcolp@sangoma.com><br/>1 Asterisk Development Team <asteriskteam@digium.com><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 George Joseph <gjoseph@digium.com><br/>1 Sylvain Afchain <safchain@gmail.com><br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Ben Ford <bford@digium.com><br/>1 lvl <digium@lvlconsultancy.nl><br/></td><td width="33%"><td width="33%">4 Ross Beer <ross.beer@voicehost.co.uk><br/>2 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Paul Brooks<br/>1 Martin Zeh<br/>1 Dmitriy Serov<br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Sylvain Afchain <safchain@wazo.io><br/>1 EDV O-TON <edv@o-ton-online.de><br/>1 Martin Zeh <martin.zeh@forsa.de><br/>1 alex <warp@adygtelecom.com><br/>1 Ross Beer<br/>1 Timothy Vanderaerden <timothy.vanderaerden@optimise-group.be><br/>1 xrobau <xrobau@gmail.com><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Paul Brooks <paul@dialaround.pro><br/>1 Peter Sokolov <newsletter@fab-online.com><br/>1 Francois Blackburn <fblackburn@wazo.io><br/>1 Peter Sokolov<br/>1 EDV O-TON<br/>1 Dmitriy Serov <serov.d.p@gmail.com><br/>1 Alex <alex@alex-at.ru><br/>1 Torrey Searle <tsearle@gmail.com><br/>1 lvl <digium@lvlconsultancy.nl><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28766">ASTERISK-28766</a>: PJSIP blind transfer not completed after using Proceeding()<br/>Reported by: lvl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b242d428051fa6e91230279caf6e10f3176c716">[4b242d4280]</a> lvl -- res_pjsip_refer: ensure refer progress is still sent after Proceeding()</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28755">ASTERISK-28755</a>: SIP/Stasis: SIP headers not transmitted in the "variables" field<br/>Reported by: Jean Aunis - Prescom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fb06121cbe0d1dc29a439dde68a36b9dc081708">[0fb06121cb]</a> Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI</li>
</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28718">ASTERISK-28718</a>: chan_sip: Returns 403 if RTP ports are depleted, should return 503<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44e1c7d93f69fbbc1b01bcb675cb5881984361bf">[44e1c7d93f]</a> Walter Doekes -- chan_sip: Return 503 if we're out of RTP ports</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28686">ASTERISK-28686</a>: chan_sip strictrtp=yes fails when media source is changed: no audio<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e99f70f112b11842e7c50c371815bc1e498ef0a6">[e99f70f112]</a> Walter Doekes -- chan_sip: Always process updated SDP on media source change</li>
</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28719">ASTERISK-28719</a>: Cannot remove defaultrule from queue using realtime queues<br/>Reported by: EDV O-TON<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3a2fbaec685d931d56f669f2d4171220e9977ac">[a3a2fbaec6]</a> Sean Bright -- res_config_odbc: Preserve empty strings returned by the database</li>
</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28755">ASTERISK-28755</a>: SIP/Stasis: SIP headers not transmitted in the "variables" field<br/>Reported by: Jean Aunis - Prescom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fb06121cbe0d1dc29a439dde68a36b9dc081708">[0fb06121cb]</a> Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28679">ASTERISK-28679</a>: stasis application is destroyed after its creation<br/>Reported by: Francois Blackburn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1051227dd96341f6e149739c1465b2e763cc67ea">[1051227dd9]</a> Kevin Harwell -- res_stasis: trigger cleanup after update</li>
</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28735">ASTERISK-28735</a>: Realtime MoH Unknown format '' -- defaulting to SLIN<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7687d1bda597ee40543f821461924f058cfaa654">[7687d1bda5]</a> Sean Bright -- res_musiconhold: Avoid spurious warning when 'format' is the empty string</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28139">ASTERISK-28139</a>: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls<br/>Reported by: Paul Brooks<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bfc93cc95428532e89ac8ffd72084dc1c811229b">[bfc93cc954]</a> Sean Bright -- chan_pjsip: Ignore RTP that we haven't negotiated</li>
</ul><br><h4>Category: Resources/res_pjsip_acl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28697">ASTERISK-28697</a>: res_pjsip: Named ACL does not update on reload if changed<br/>Reported by: Timothy Vanderaerden<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a60071610c490e3a63f2eeab072d45e2b22d04f1">[a60071610c]</a> Joshua C. Colp -- pjsip: Update ACLs on named ACL changes.</li>
</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26082">ASTERISK-26082</a>: res_pjsip_messaging: MessageSend Content-Type can't be changed<br/>Reported by: Alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dafdcc623cd46dba3663a3b5bb2663f8630a3a00">[dafdcc623c]</a> Sean Bright -- res_pjsip_messaging: Allow Content-Type to be overridden</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25421">ASTERISK-25421</a>: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d99580ae69eb31baa8fc2b05e681320e24fcff9c">[d99580ae69]</a> Sean Bright -- res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28714">ASTERISK-28714</a>: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e14fd3b22a832f1e49fbdf340727c12f0c0d2929">[e14fd3b22a]</a> Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28754">ASTERISK-28754</a>: ASTERISK-28738 Causes Audio Issue After Hold<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed670ca0c666902cc066a6c557fe3c0baaa389ea">[ed670ca0c6]</a> Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH transitions</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28738">ASTERISK-28738</a>: Incorrect state machine used when MOH_PASSTHRU is used<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21ed67189332cd527718bd2a6d25512bf867accb">[21ed671893]</a> Torrey Searle -- res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28716">ASTERISK-28716</a>: ICE: pjnath shouldn't wait for ICE to complete before allowing sending<br/>Reported by: Benjamin Keith Ford<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f928600a98dde613e0bdebf8c24bc4e8b08e822d">[f928600a98]</a> Ben Ford -- RTP/ICE: Send on first valid pair.</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78d9bc8ef66e2d20925daf0eb597cf8a956e1029">[78d9bc8ef6]</a> Kevin Harwell -- stasis/app: don't lock an app before a call to send</li>
</ul><br><h4>Category: Utilities/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28685">ASTERISK-28685</a>: check_expr2: linking (when hardening) and cross-compiling troubles<br/>Reported by: Sebastian Kemper<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d6b5a18111e1dff06b623ce3e92d771b46d59b7e">[d6b5a18111]</a> Sebastian Kemper -- check_expr2: fix cross-compile/hardening issues</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26955">ASTERISK-26955</a>: pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected<br/>Reported by: Peter Sokolov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5fa23e72af535af41e91b4d063a4f90178f33a5">[d5fa23e72a]</a> Sean Bright -- pjproject_bundled: Allow brackets in via parameters</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24798">ASTERISK-24798</a>: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor<br/>Reported by: xrobau<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0246bca5817a808316483ce399976a024b9cdcfb">[0246bca581]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28726">ASTERISK-28726</a>: install_prereq script uses the interactive mode when installing aptitude<br/>Reported by: Sylvain Afchain<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9419295aeefd0fc0f3a6c248c3f514770173d876">[9419295aee]</a> Sylvain Afchain -- install_prereq: Install aptitude non-interactively</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28750">ASTERISK-28750</a>: TLS/SSL Key too small error<br/>Reported by: Martin Zeh<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cbbf192f0aac2c24f3007ba3d9597bc28d05af70">[cbbf192f0a]</a> Sean Bright -- tcptls.c: Log more informative OpenSSL errors</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24798">ASTERISK-24798</a>: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor<br/>Reported by: xrobau<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0246bca5817a808316483ce399976a024b9cdcfb">[0246bca581]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
</ul><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><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=077cc24d6610425aeb12bbe1fb2fdffb1c345088">077cc24d66</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 13.32.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3e9d37ce78960729b91968dfeb6df4fb1c3a0b9">c3e9d37ce7</a></td><td>Walter Doekes</td><td>say: Remove unused "plural" option from main/say</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6995b0a8fa472d40c4d9a5704ce2a626684d6b85">6995b0a8fa</a></td><td>Walter Doekes</td><td>app_queue: Refactor odd placement of if's around say_position</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d13cd25d4c21d1135b48cd97117631c685061271">d13cd25d4c</a></td><td>Kevin Harwell</td><td>format_cap: make function parameters 'const'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ff705728748a803bc2072b3ed1e2a9b959aff95">2ff7057287</a></td><td>Jaco Kroon</td><td>addons/res_config_mysql: silense warnings about printf format errors.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=66388abbb18cb1f9f5b22cab72795950bab3e106">66388abbb1</a></td><td>Sean Bright</td><td>ast_tls_cert: Allow private key size to be set on command line</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea1f318534ffc322d125ee662c61b1ec73262431">ea1f318534</a></td><td>Sean Bright</td><td>func_odbc: Prevent snprintf() truncation warning</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74d0244dc3b25a11dbcdcaa25ccc9036027485c3">74d0244dc3</a></td><td>George Joseph</td><td>doc: Fix CHANGES entries to have .txt suffix and update READMEs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cea6cad34383448e71cc42c33b466509bc9d0896">cea6cad343</a></td><td>Walter Doekes</td><td>chan_sip: Clarify in sample docs how directmediapermit/-acl should be used</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><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>.lastclean | 1
.version | 1
ChangeLog |68939 ----------
asterisk-13.31.0-summary.html | 188
asterisk-13.31.0-summary.txt | 578
b/CHANGES | 25
b/UPGRADE.txt | 56
b/addons/res_config_mysql.c | 16
b/apps/app_mixmonitor.c | 29
b/apps/app_queue.c | 60
b/channels/chan_pjsip.c | 3
b/channels/chan_sip.c | 126
b/channels/sip/include/sip.h | 1
b/configs/samples/sip.conf.sample | 4
b/configure | 141
b/configure.ac | 22
b/contrib/scripts/ast_tls_cert | 8
b/contrib/scripts/install_prereq | 2
b/doc/CHANGES-staging/README.md | 8
b/doc/UPGRADE-staging/README.md | 7
b/funcs/func_odbc.c | 4
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format_cap.h | 4
b/include/asterisk/message.h | 13
b/include/asterisk/res_pjsip_session.h | 4
b/include/asterisk/say.h | 4
b/include/asterisk/sorcery.h | 27
b/main/file.c | 2
b/main/format_cap.c | 4
b/main/message.c | 21
b/main/say.c | 10
b/main/sorcery.c | 46
b/main/tcptls.c | 26
b/makeopts.in | 2
b/menuselect/configure | 22
b/res/ari/ari_model_validators.c | 59
b/res/ari/ari_model_validators.h | 23
b/res/res_config_odbc.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip/pjsip_configuration.c | 19
b/res/res_pjsip_acl.c | 20
b/res/res_pjsip_messaging.c | 54
b/res/res_pjsip_refer.c | 7
b/res/res_pjsip_sdp_rtp.c | 53
b/res/res_pjsip_session.c | 5
b/res/res_rtp_asterisk.c | 33
b/res/res_sorcery_config.c | 1
b/res/stasis/messaging.c | 11
b/rest-api/api-docs/endpoints.json | 20
b/rest-api/resources.json | 2
b/third-party/pjproject/configure.m4 | 1
b/third-party/pjproject/patches/0040-ICE-Add-callback-for-finding-valid-pair.patch | 84
b/third-party/pjproject/patches/0040-brackets-in-via-received-params.patch | 6
contrib/realtime/mysql/mysql_cdr.sql | 33
contrib/realtime/mysql/mysql_config.sql | 1189
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 37
contrib/realtime/postgresql/postgresql_config.sql | 1287
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
doc/CHANGES-staging/res_fax_negotiate_both | 7
60 files changed, 787 insertions(+), 72651 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.32.0-rc1
Date: 2020-03-05
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version 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-13.31.0.
----------------------------------------------------------------------
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 affected by commits that went into
this release.
Coders Testers Reporters
10 Sean Bright 4 Ross Beer
5 Walter Doekes 2 Walter Doekes
4 Kevin Harwell 1 Paul Brooks
2 Torrey Searle 1 Martin Zeh
2 Joshua C. Colp 1 Dmitriy Serov
1 Asterisk Development Team 1 Jean Aunis - Prescom
1 Sebastian Kemper 1 Benjamin Keith Ford
1 George Joseph 1 Sylvain Afchain
1 Sylvain Afchain 1 EDV O-TON
1 Jaco Kroon 1 Martin Zeh
1 Ben Ford 1 alex
1 lvl 1 Ross Beer
1 Timothy Vanderaerden
1 xrobau
1 Sebastian Kemper
1 Paul Brooks
1 Peter Sokolov
1 Francois Blackburn
1 Peter Sokolov
1 EDV O-TON
1 Dmitriy Serov
1 Alex
1 Torrey Searle
1 lvl
----------------------------------------------------------------------
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.
Bug
Category: Channels/chan_pjsip
ASTERISK-28766: PJSIP blind transfer not completed after using
Proceeding()
Reported by: lvl
* [4b242d4280] lvl -- res_pjsip_refer: ensure refer progress is still
sent after Proceeding()
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables"
field
Reported by: Jean Aunis - Prescom
* [0fb06121cb] Kevin Harwell -- message & stasis/messaging: make text
message variables work in ARI
Category: Channels/chan_sip/Interoperability
ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should
return 503
Reported by: Walter Doekes
* [44e1c7d93f] Walter Doekes -- chan_sip: Return 503 if we're out of RTP
ports
ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed:
no audio
Reported by: Walter Doekes
* [e99f70f112] Walter Doekes -- chan_sip: Always process updated SDP on
media source change
Category: Core/Configuration
ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues
Reported by: EDV O-TON
* [a3a2fbaec6] Sean Bright -- res_config_odbc: Preserve empty strings
returned by the database
Category: Core/Stasis
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables"
field
Reported by: Jean Aunis - Prescom
* [0fb06121cb] Kevin Harwell -- message & stasis/messaging: make text
message variables work in ARI
Category: Resources/res_ari
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
* [1051227dd9] Kevin Harwell -- res_stasis: trigger cleanup after update
Category: Resources/res_musiconhold
ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN
Reported by: Ross Beer
* [7687d1bda5] Sean Bright -- res_musiconhold: Avoid spurious warning
when 'format' is the empty string
Category: Resources/res_pjsip
ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop
Calls
Reported by: Paul Brooks
* [bfc93cc954] Sean Bright -- chan_pjsip: Ignore RTP that we haven't
negotiated
Category: Resources/res_pjsip_acl
ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed
Reported by: Timothy Vanderaerden
* [a60071610c] Joshua C. Colp -- pjsip: Update ACLs on named ACL
changes.
Category: Resources/res_pjsip_messaging
ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be
changed
Reported by: Alex
* [dafdcc623c] Sean Bright -- res_pjsip_messaging: Allow Content-Type to
be overridden
ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
error when sending
Reported by: Dmitriy Serov
* [d99580ae69] Sean Bright -- res_pjsip_messaging: Ensure
MESSAGE_SEND_STATUS is set properly
Category: Resources/res_pjsip_pubsub
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate
(ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
* [e14fd3b22a] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence
data ref when recreating.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold
Reported by: Ross Beer
* [ed670ca0c6] Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH
transitions
ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used
Reported by: Torrey Searle
* [21ed671893] Torrey Searle -- res_pjsip_sdp_rtp: implement hold state
handling on moh_passthrough
Category: Resources/res_rtp_asterisk
ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before
allowing sending
Reported by: Benjamin Keith Ford
* [f928600a98] Ben Ford -- RTP/ICE: Send on first valid pair.
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
* [78d9bc8ef6] Kevin Harwell -- stasis/app: don't lock an app before a
call to send
Category: Utilities/General
ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling
troubles
Reported by: Sebastian Kemper
* [d6b5a18111] Sebastian Kemper -- check_expr2: fix
cross-compile/hardening issues
Category: pjproject/pjsip
ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6
Address Delimited by "[]" Rejected
Reported by: Peter Sokolov
* [d5fa23e72a] Sean Bright -- pjproject_bundled: Allow brackets in via
parameters
Improvement
Category: Applications/app_mixmonitor
ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name
Extension In MixMonitor
Reported by: xrobau
* [0246bca581] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to
correct value when wav49 is used
Category: Contrib/General
ASTERISK-28726: install_prereq script uses the interactive mode when
installing aptitude
Reported by: Sylvain Afchain
* [9419295aee] Sylvain Afchain -- install_prereq: Install aptitude
non-interactively
Category: Core/HTTP
ASTERISK-28750: TLS/SSL Key too small error
Reported by: Martin Zeh
* [cbbf192f0a] Sean Bright -- tcptls.c: Log more informative OpenSSL
errors
Category: Documentation
ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name
Extension In MixMonitor
Reported by: xrobau
* [0246bca581] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to
correct value when wav49 is used
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+------------------+----------------------------------------|
| 077cc24d66 | Asterisk | Update CHANGES and UPGRADE.txt for |
| | Development Team | 13.32.0 |
|------------+------------------+----------------------------------------|
| c3e9d37ce7 | Walter Doekes | say: Remove unused "plural" option |
| | | from main/say |
|------------+------------------+----------------------------------------|
| 6995b0a8fa | Walter Doekes | app_queue: Refactor odd placement of |
| | | if's around say_position |
|------------+------------------+----------------------------------------|
| d13cd25d4c | Kevin Harwell | format_cap: make function parameters |
| | | 'const' |
|------------+------------------+----------------------------------------|
| 2ff7057287 | Jaco Kroon | addons/res_config_mysql: silense |
| | | warnings about printf format errors. |
|------------+------------------+----------------------------------------|
| 66388abbb1 | Sean Bright | ast_tls_cert: Allow private key size |
| | | to be set on command line |
|------------+------------------+----------------------------------------|
| ea1f318534 | Sean Bright | func_odbc: Prevent snprintf() |
| | | truncation warning |
|------------+------------------+----------------------------------------|
| 74d0244dc3 | George Joseph | doc: Fix CHANGES entries to have .txt |
| | | suffix and update READMEs |
|------------+------------------+----------------------------------------|
| cea6cad343 | Walter Doekes | chan_sip: Clarify in sample docs how |
| | | directmediapermit/-acl should be used |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
.lastclean | 1
.version | 1
ChangeLog |68939 ----------
asterisk-13.31.0-summary.html | 188
asterisk-13.31.0-summary.txt | 578
b/CHANGES | 25
b/UPGRADE.txt | 56
b/addons/res_config_mysql.c | 16
b/apps/app_mixmonitor.c | 29
b/apps/app_queue.c | 60
b/channels/chan_pjsip.c | 3
b/channels/chan_sip.c | 126
b/channels/sip/include/sip.h | 1
b/configs/samples/sip.conf.sample | 4
b/configure | 141
b/configure.ac | 22
b/contrib/scripts/ast_tls_cert | 8
b/contrib/scripts/install_prereq | 2
b/doc/CHANGES-staging/README.md | 8
b/doc/UPGRADE-staging/README.md | 7
b/funcs/func_odbc.c | 4
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format_cap.h | 4
b/include/asterisk/message.h | 13
b/include/asterisk/res_pjsip_session.h | 4
b/include/asterisk/say.h | 4
b/include/asterisk/sorcery.h | 27
b/main/file.c | 2
b/main/format_cap.c | 4
b/main/message.c | 21
b/main/say.c | 10
b/main/sorcery.c | 46
b/main/tcptls.c | 26
b/makeopts.in | 2
b/menuselect/configure | 22
b/res/ari/ari_model_validators.c | 59
b/res/ari/ari_model_validators.h | 23
b/res/res_config_odbc.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip/pjsip_configuration.c | 19
b/res/res_pjsip_acl.c | 20
b/res/res_pjsip_messaging.c | 54
b/res/res_pjsip_refer.c | 7
b/res/res_pjsip_sdp_rtp.c | 53
b/res/res_pjsip_session.c | 5
b/res/res_rtp_asterisk.c | 33
b/res/res_sorcery_config.c | 1
b/res/stasis/messaging.c | 11
b/rest-api/api-docs/endpoints.json | 20
b/rest-api/resources.json | 2
b/third-party/pjproject/configure.m4 | 1
b/third-party/pjproject/patches/0040-ICE-Add-callback-for-finding-valid-pair.patch | 84
b/third-party/pjproject/patches/0040-brackets-in-via-received-params.patch | 6
contrib/realtime/mysql/mysql_cdr.sql | 33
contrib/realtime/mysql/mysql_config.sql | 1189
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 37
contrib/realtime/postgresql/postgresql_config.sql | 1287
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
doc/CHANGES-staging/res_fax_negotiate_both | 7
60 files changed, 787 insertions(+), 72651 deletions(-)

View File

@@ -0,0 +1,33 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start DATETIME,
answer DATETIME,
end DATETIME,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,35 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BLOB,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';

View File

@@ -0,0 +1,37 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start TIMESTAMP WITHOUT TIME ZONE,
answer TIMESTAMP WITHOUT TIME ZONE,
"end" TIMESTAMP WITHOUT TIME ZONE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
COMMIT;

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,39 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BYTEA,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
COMMIT;