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Update for 20.1.0

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Asterisk Development Team 2023-01-12 06:46:01 -05:00
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20.1.0-rc2
20.1.0

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2023-01-12 11:46 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 20.1.0 Released.
2023-01-05 19:15 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 20.1.0-rc2 Released.

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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-20.1.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-20.1.0-rc2</h3><h3 align="center">Date: 2023-01-05</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="#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-20.1.0-rc1.</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%">1 George Joseph <gjoseph@digium.com><br/></td><td width="33%"><td width="33%">1 Joshua C. Colp <jcolp@digium.com><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: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30369">ASTERISK-30369</a>: res_pjsip: Websockets from same IP shut down when they shouldn't be<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff9ab3857392ea58624e2f40b8a5c35e4224ced6">[ff9ab38573]</a> George Joseph -- res_pjsip_transport_websocket: Add remote port to transport</li>
</ul><br><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>0 files changed</pre><br></html>

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Release Summary
asterisk-20.1.0-rc2
Date: 2023-01-05
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. 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-20.1.0-rc1.
----------------------------------------------------------------------
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
1 George Joseph 1 Joshua C. Colp
----------------------------------------------------------------------
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: Resources/res_pjsip
ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they
shouldn't be
Reported by: Joshua C. Colp
* [ff9ab38573] George Joseph -- res_pjsip_transport_websocket: Add
remote port to transport
----------------------------------------------------------------------
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.
0 files changed

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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-20.1.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-20.1.0</h3><h3 align="center">Date: 2023-01-12</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="#open_issues">Open 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-20.0.0.</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%">39 Naveen Albert <asterisk@phreaknet.org><br/>7 George Joseph <gjoseph@digium.com><br/>5 Philip Prindeville <philipp@redfish-solutions.com><br/>4 Asterisk Development Team <asteriskteam@digium.com><br/>4 Mike Bradeen <mbradeen@sangoma.com><br/>3 Maximilian Fridrich <m.fridrich@commend.com><br/>3 Joshua C. Colp <jcolp@sangoma.com><br/>2 Jaco Kroon <jaco@uls.co.za><br/>2 Michael Kuron <m.kuron@gmx.de><br/>2 Marcel Wagner <mwagner@sipgate.de><br/>1 Birger Harzenetter (license 5870)<br/>1 Holger Hans Peter Freyther <holger@moiji-mobile.com><br/>1 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Alexandre Fournier <afournier@wazo.io><br/>1 Ben Ford <bford@digium.com><br/>1 Henning Westerholt <hw@gilawa.com><br/></td><td width="33%">1 inspecting SIP traces.<br/></td><td width="33%">37 N A <asterisk@phreaknet.org><br/>7 George Joseph <gjoseph@digium.com><br/>4 Philip Prindeville <philipp@redfish-solutions.com><br/>3 Matt Jordan <mjordan@digium.com><br/>3 Maximilian Fridrich <m.fridrich@commend.com><br/>2 Michael Bradeen <mbradeen@sangoma.com><br/>2 Marcel Wagner <mwagner@sipgate.de><br/>2 Joshua C. Colp <jcolp@digium.com><br/>1 Igor Yeroshev <igor.yeroshev@kaspi.kz><br/>1 David Uczen <duczen@fluentstream.com><br/>1 Alexandre Fournier <afournier@wazo.io><br/>1 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 shawty <shawty.d.ds@googlemail.com><br/>1 David Woolley <forums@david-woolley.me.uk><br/>1 nappsoft <infos@nappsoft.ch><br/>1 Jesse Ross <jesseross92@gmail.com><br/>1 Joeran Vinzens <vinzens@sipgate.de><br/>1 Henning Westerholt <hw@gilawa.com><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Saken <saken.tastemirov@kaspi.kz><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>Security</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30103">ASTERISK-30103</a>: chan_ooh323 Vulnerability in calling/called party IE<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb1d7ab53cf8ba8c42432cdedd5943102efe5101">[eb1d7ab53c]</a> Mike Bradeen -- ooh323c: not checking for IE minimum length</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30176">ASTERISK-30176</a>: manager: GetConfig can read files outside of Asterisk<br/>Reported by: shawty<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=81f10e847efdbe8ec264062ee234e1098c29b3f6">[81f10e847e]</a> Mike Bradeen -- manager: prevent file access outside of config dir</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30338">ASTERISK-30338</a>: pjproject: Backport security fixes from 2.13<br/>Reported by: Benjamin Keith Ford<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4769947688a54ac926240ed4a68d16a3d1807ba">[d476994768]</a> Ben Ford -- pjproject: 2.13 security fixes</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30286">ASTERISK-30286</a>: app_mixmonitor: Add option to use real Caller ID for Caller ID<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=691178c48e58923b7c3b122f90a663a83d9695e1">[691178c48e]</a> Naveen Albert -- app_mixmonitor: Add option to use real Caller ID for voicemail.</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30223">ASTERISK-30223</a>: features: add no-answer option to Bridge application<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5ec60e6c61fd5addb8a2381e944d4a4a42e985e">[a5ec60e6c6]</a> Naveen Albert -- features: Add no answer option to Bridge.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30290">ASTERISK-30290</a>: file.c: Don't emit warnings on winks.<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8142b313c36e96f46c29938ded8fca5494064a80">[8142b313c3]</a> Naveen Albert -- file.c: Don't emit warnings on winks.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30327">ASTERISK-30327</a>: rtp_engine.h: Remove obsolete example usage<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a4bcdce1db1aa411b6895456dc939800837437d3">[a4bcdce1db]</a> Naveen Albert -- rtp_engine.h: Update examples using ast_format_set.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30328">ASTERISK-30328</a>: Typo in from_domain description on res_pjsip configuration documentation<br/>Reported by: Marcel Wagner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=58534b309fe0aa212c32c85f1c0497b0738a624b">[58534b309f]</a> Marcel Wagner -- res_pjsip: Fix typo in from_domain documentation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30316">ASTERISK-30316</a>: res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations<br/>Reported by: Marcel Wagner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b83af13f65ba271c041e18fd344bb0b057aa00bc">[b83af13f65]</a> Marcel Wagner -- res_pjsip: Update contact_user to point out default</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30289">ASTERISK-30289</a>: xmldoc: Allow XML docs to be reloaded<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52c7d3ed07a744a2f2a199ba58e3a9896b63af19">[52c7d3ed07]</a> Naveen Albert -- xmldoc: Allow XML docs to be reloaded.</li>
</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30308">ASTERISK-30308</a>: pbx_builtins: Allow Answer to return immediately<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c7df5ee7c134606e1dd9b255f4d67012c09b0f8c">[c7df5ee7c1]</a> Naveen Albert -- pbx_builtins: Allow Answer to return immediately.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30158">ASTERISK-30158</a>: PJSIP: Add new 100rel option "peer_supported"<br/>Reported by: Maximilian Fridrich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5bbad0d27c967a28a5824ae10fdca714e064c46d">[5bbad0d27c]</a> Maximilian Fridrich -- res_pjsip: Add 100rel option "peer_supported".</li>
</ul><br><h4>Category: Resources/res_pjsip_geolocation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30241">ASTERISK-30241</a>: res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8afb313a4307dbff8849b036703775d8770d21a9">[8afb313a43]</a> Naveen Albert -- res_pjsip_geolocation: Change some notices to debugs.</li>
</ul><br><h4>Category: Tests/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30295">ASTERISK-30295</a>: test_json: Remove duplicated static function<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e77b8c473c624e0d8c1a1c1ea8ab455d89c7e34">[1e77b8c473]</a> Naveen Albert -- test_json: Remove duplicated static function.</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_if</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30349">ASTERISK-30349</a>: app_if: Format truncation error<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f9cdfbc508fe0c1defd2743cee234c424571388">[2f9cdfbc50]</a> Naveen Albert -- app_if: Fix format truncation errors.</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29998">ASTERISK-29998</a>: sla: deadlock when calling SLAStation application<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2efa290d3c96cbcfa6095b1361863f7bc19b14a6">[2efa290d3c]</a> Naveen Albert -- sla: Prevent deadlock and crash due to autoservicing.</li>
</ul><br><h4>Category: Applications/app_saynumber</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30248">ASTERISK-30248</a>: ast_get_digit_str adds bogus initial delimiter if first character not to be spoken<br/>Reported by: David Woolley<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=407216a0a56afdd692be4f7253ce710eb10d02d2">[407216a0a5]</a> Naveen Albert -- say: Don't prepend ampersand erroneously.</li>
</ul><br><h4>Category: Applications/app_stack</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30076">ASTERISK-30076</a>: app_stack: Incorrect exit location in predial handlers logged<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dfe2f38642099a8797037964c7dff7e15abdff56">[dfe2f38642]</a> Naveen Albert -- app_stack: Print proper exit location for PBXless channels.</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30283">ASTERISK-30283</a>: app_voicemail: Fix msg_create_from_file not sending email to user<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9c031c1f82244e00fddfc9f5e8f0d83e5548b34">[b9c031c1f8]</a> Naveen Albert -- app_voicemail: Fix missing email in msg_create_from_file.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30248">ASTERISK-30248</a>: ast_get_digit_str adds bogus initial delimiter if first character not to be spoken<br/>Reported by: David Woolley<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=407216a0a56afdd692be4f7253ce710eb10d02d2">[407216a0a5]</a> Naveen Albert -- say: Don't prepend ampersand erroneously.</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30336">ASTERISK-30336</a>: sig_analog: Fix no timeout duration<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b90e57758befa4048f3a56493e9ee23afec7eb6b">[b90e57758b]</a> Naveen Albert -- sig_analog: Fix no timeout duration.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30274">ASTERISK-30274</a>: chan_dahdi: Unavailable channels are BUSY<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=180ca32565efbed902463e06d9ee200d4dc25a06">[180ca32565]</a> Naveen Albert -- chan_dahdi: Fix unavailable channels returning busy.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30256">ASTERISK-30256</a>: chan_dahdi: Fix format truncation warnings<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98fc05f13b0b245d7a5a077153da47229d142323">[98fc05f13b]</a> Naveen Albert -- chan_dahdi: Resolve format truncation warning.</li>
</ul><br><h4>Category: Channels/chan_local</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30239">ASTERISK-30239</a>: Prometheus plugin crashes Asterisk when using local channel<br/>Reported by: Joeran Vinzens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62881c668b0e7cb097e2c36f0e88d18dcd1aa38d">[62881c668b]</a> Holger Hans Peter Freyther -- res_prometheus: Do not crash on invisible bridges</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30184">ASTERISK-30184</a>: res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg<br/>Reported by: Maximilian Fridrich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60b81eabe080af842d81b9d84d1dcedb24b87398">[60b81eabe0]</a> Maximilian Fridrich -- core & res_pjsip: Improve topology change handling.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30193">ASTERISK-30193</a>: chan_pjsip should return all codecs on a re-INVITE without SDP<br/>Reported by: Henning Westerholt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=12445040d346805b6b63a171e07855f7662881af">[12445040d3]</a> Henning Westerholt -- res_pjsip: return all codecs on a re-INVITE without SDP</li>
</ul><br><h4>Category: Channels/chan_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30281">ASTERISK-30281</a>: chan_rtp: Local address being used before being set<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f723b465e5baf4c625a986ae7b6513585d15c44a">[f723b465e5]</a> George Joseph -- chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer</li>
</ul><br><h4>Category: Core/AstDB</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30245">ASTERISK-30245</a>: db: ListItems is incorrect<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7335b0cffe6c997eb66de249fd90c7463abd5828">[7335b0cffe]</a> Birger Harzenetter -- db: Fix incorrect DB tree count for AMI.</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30321">ASTERISK-30321</a>: Build: Embedded blobs have executable stacks<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ce2153fc5aa4e1b252d4d9d608748b6615460f30">[ce2153fc5a]</a> Jaco Kroon -- Build system: Avoid executable stack.</li>
</ul><br><h4>Category: Core/Dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30258">ASTERISK-30258</a>: Dialing API: Cancel a running async thread, does not always cancel all calls<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50a4495799fd4e5f15da5695a034b05943858f02">[50a4495799]</a> Frederic LE FOLL -- Dialing API: Cancel a running async thread, may not cancel all calls</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30278">ASTERISK-30278</a>: tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afd86b47c1e886638454abf041ffa50d914b5c67">[afd86b47c1]</a> Naveen Albert -- tcptls: Prevent crash when freeing OpenSSL errors.</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30278">ASTERISK-30278</a>: tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afd86b47c1e886638454abf041ffa50d914b5c67">[afd86b47c1]</a> Naveen Albert -- tcptls: Prevent crash when freeing OpenSSL errors.</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30285">ASTERISK-30285</a>: manager.c: Remove outdated documentation<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49cfdbbdff315e5a6bb665bb44a09d881f71b52c">[49cfdbbdff]</a> Naveen Albert -- manager: Update ModuleCheck documentation.</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30293">ASTERISK-30293</a>: Memory leak in JSON_DECODE<br/>Reported by: David Uczen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=002afc3f2a4139fae0049c1d9e2fb591ca8ef040">[002afc3f2a]</a> Naveen Albert -- func_json: Fix memory leak.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30220">ASTERISK-30220</a>: func_scramble: Fix segfault due to null pointer deref<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8aae0b9f083af6de2ad2dfc87d2f1eaf4e20f4a0">[8aae0b9f08]</a> Naveen Albert -- func_scramble: Fix null pointer dereference.</li>
</ul><br><h4>Category: Functions/func_logic</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30243">ASTERISK-30243</a>: func_logic: IF function complains if both branches are empty<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=407167cc283ee5cf53ccabe2699f9610b3b768ea">[407167cc28]</a> Naveen Albert -- func_logic: Don't emit warning if both IF branches are empty.</li>
</ul><br><h4>Category: Functions/func_presencestate</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30311">ASTERISK-30311</a>: func_presencestate: Fix invalid memory access.<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=83eb113e0f31fa328a85233283854c985c42b064">[83eb113e0f]</a> Naveen Albert -- func_presencestate: Fix invalid memory access.</li>
</ul><br><h4>Category: Resources/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30239">ASTERISK-30239</a>: Prometheus plugin crashes Asterisk when using local channel<br/>Reported by: Joeran Vinzens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62881c668b0e7cb097e2c36f0e88d18dcd1aa38d">[62881c668b]</a> Holger Hans Peter Freyther -- res_prometheus: Do not crash on invisible bridges</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30237">ASTERISK-30237</a>: res_prometheus: Crash when scraping bridges<br/>Reported by: Igor Yeroshev<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62881c668b0e7cb097e2c36f0e88d18dcd1aa38d">[62881c668b]</a> Holger Hans Peter Freyther -- res_prometheus: Do not crash on invisible bridges</li>
</ul><br><h4>Category: Resources/res_adsi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29793">ASTERISK-29793</a>: adsi: CAS is malformed<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=80e6205bb05f920022b959685155c3b3ce1668ec">[80e6205bb0]</a> Naveen Albert -- res_adsi: Fix major regression caused by media format rearchitecture.</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30314">ASTERISK-30314</a>: res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61922d293435dc8b5724d1fddb02f47e75d1608a">[61922d2934]</a> Joshua C. Colp -- res_agi: Respect "transmit_silence" option for "RECORD FILE".</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30344">ASTERISK-30344</a>: ari: Memory leak in create when specifying JSON<br/>Reported by: Saken<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=564349ff5d4f39ee51b69655babea377e3edb065">[564349ff5d]</a> Joshua C. Colp -- ari: Destroy body variables in channel create.</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30237">ASTERISK-30237</a>: res_prometheus: Crash when scraping bridges<br/>Reported by: Igor Yeroshev<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62881c668b0e7cb097e2c36f0e88d18dcd1aa38d">[62881c668b]</a> Holger Hans Peter Freyther -- res_prometheus: Do not crash on invisible bridges</li>
</ul><br><h4>Category: Resources/res_crypto</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30213">ASTERISK-30213</a>: Make crypto_load() reentrant and handle symlinks correctly<br/>Reported by: Philip Prindeville<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d0bea5a725ecdef00536da8380a805f11b8cefc4">[d0bea5a725]</a> Philip Prindeville -- res_crypto: handle unsafe private key files</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ef74ecacc7ffd81a505154f4590df12b913eed55">[ef74ecacc7]</a> Philip Prindeville -- res_crypto: don't modify fname in try_load_key()</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e2485b5c0d4a80015ba6f6107f28e3d2bd6480b">[5e2485b5c0]</a> Philip Prindeville -- res_crypto: use ast_file_read_dirs() to iterate</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30235">ASTERISK-30235</a>: res_crypto and tests: Memory issues and and uninitialized variable error<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e25b690d1023927f092030d9b743338c5a0e2223">[e25b690d10]</a> George Joseph -- res_crypto: Memory issues and uninitialized variable errors</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30226">ASTERISK-30226</a>: REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=026dc08529254f7088aa77df3f417d6d6304b6b6">[026dc08529]</a> Philip Prindeville -- res_crypto: don't complain about directories</li>
</ul><br><h4>Category: Resources/res_geolocation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30234">ASTERISK-30234</a>: res_geolocation: ...may be used uninitialized error in geoloc_config.c<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e33f2dcc0f37ab7ed294dcd4473ba4735955edcf">[e33f2dcc0f]</a> George Joseph -- res_geolocation: Fix issues exposed by compiling with -O2</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30369">ASTERISK-30369</a>: res_pjsip: Websockets from same IP shut down when they shouldn't be<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff9ab3857392ea58624e2f40b8a5c35e4224ced6">[ff9ab38573]</a> George Joseph -- res_pjsip_transport_websocket: Add remote port to transport</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30184">ASTERISK-30184</a>: res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg<br/>Reported by: Maximilian Fridrich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60b81eabe080af842d81b9d84d1dcedb24b87398">[60b81eabe0]</a> Maximilian Fridrich -- core & res_pjsip: Improve topology change handling.</li>
</ul><br><h4>Category: Resources/res_pjsip_exten_state</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30264">ASTERISK-30264</a>: res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9258d8212a1fd15bd53ab998a7dda709000f1f93">[9258d8212a]</a> Naveen Albert -- res_pjsip_pubsub: Prevent removing subscriptions.</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30264">ASTERISK-30264</a>: res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9258d8212a1fd15bd53ab998a7dda709000f1f93">[9258d8212a]</a> Naveen Albert -- res_pjsip_pubsub: Prevent removing subscriptions.</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30217">ASTERISK-30217</a>: Registration do not allow multiple proxies<br/>Reported by: Igor Goncharovsky<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=096529d33fe65a893f6abb420fd4291c36c59be6">[096529d33f]</a> Igor Goncharovsky -- res_pjsip_outbound_registration: Allow to use multiple proxies for registration</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30244">ASTERISK-30244</a>: res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed<br/>Reported by: nappsoft<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7684c9e907fb85f5c58b025d9e385ad2600f12a2">[7684c9e907]</a> George Joseph -- pjsip_transport_events: Fix possible use after free on transport</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30265">ASTERISK-30265</a>: res_pjsip_session: Fix missing PLAR support on INVITEs<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d6003fa9a2afb21c8953d5ac7238ea83412086d">[0d6003fa9a]</a> Naveen Albert -- res_pjsip_session.c: Map empty extensions in INVITEs to s.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30184">ASTERISK-30184</a>: res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg<br/>Reported by: Maximilian Fridrich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60b81eabe080af842d81b9d84d1dcedb24b87398">[60b81eabe0]</a> Maximilian Fridrich -- core & res_pjsip: Improve topology change handling.</li>
</ul><br><h4>Category: Tests/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30282">ASTERISK-30282</a>: CI: Coredump output isn't saved when running unittests<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c1c623deed2d8318d3bbd55af810d4243272899">[0c1c623dee]</a> George Joseph -- runUnittests.sh: Save coredumps to proper directory</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30232">ASTERISK-30232</a>: Initialize stack-based ast_test_capture structures correctly<br/>Reported by: Philip Prindeville<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e7ce90f9c44e1fbd7bdfdb83424c0149eb555d9">[3e7ce90f9c]</a> Philip Prindeville -- test: initialize capture structure before freeing</li>
</ul><br><h4>Category: Tests/Portability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30273">ASTERISK-30273</a>: test_mwi: compilation fails on 32-bit Debian<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ca8900b0f62edf83b39810c58f7b2445a85b776a">[ca8900b0f6]</a> Naveen Albert -- tests: Fix compilation errors on 32-bit.</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28689">ASTERISK-28689</a>: res_pjsip: Crash when locking group lock when sending stateful response<br/>Reported by: Jesse Ross<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50e2921a48c8c3b4bd008aab7d9dc996bcdcced6">[50e2921a48]</a> Mike Bradeen -- res_pjsip: prevent crash on websocket disconnect</li>
</ul><br><h3>New Feature</h3><h4>Category: Applications/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29497">ASTERISK-29497</a>: Add conditional branch applications<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b365ea86013231c87b7c9b2c0868068e0f68c23b">[b365ea8601]</a> Naveen Albert -- app_if: Adds conditional branch applications</li>
</ul><br><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30179">ASTERISK-30179</a>: app_amd: Allow audio to be played while AMD is running<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c791f9a656434386136669fb8ae19ea746cf938">[8c791f9a65]</a> Naveen Albert -- app_amd: Add option to play audio during AMD.</li>
</ul><br><h4>Category: Applications/app_bridgewait</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30216">ASTERISK-30216</a>: app_bridgewait: Add option for BridgeWait to not answer<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e29607b5c66ad45f586069867f649cd2232fd63">[1e29607b5c]</a> Naveen Albert -- app_bridgewait: Add option to not answer channel.</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30284">ASTERISK-30284</a>: app_mixmonitor: Add option to delete recording file when done<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6e59b01e1a1e4b059df6131b56079ca7a2abf7ca">[6e59b01e1a]</a> Naveen Albert -- app_mixmonitor: Add option to delete files on exit.</li>
</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30091">ASTERISK-30091</a>: cdr: Allow CDRs to ignore call state changes<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b331caca30ca8f97011c8045afa2419e1f9243e5">[b331caca30]</a> Naveen Albert -- cdr: Allow bridging and dial state changes to be ignored.</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30305">ASTERISK-30305</a>: chan_dahdi: Allow FXO channels to start immediately<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5ede4e217a8afc179412db2d067ee552d175b97a">[5ede4e217a]</a> Naveen Albert -- chan_dahdi: Allow FXO channels to start immediately.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21502">ASTERISK-21502</a>: New SIP Channel Driver - add Advice of Charge support<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b6855755ce2ca249dc87159ca92aa7be6152712f">[b6855755ce]</a> Joshua C. Colp -- res_pjsip_aoc: Don't assume a body exists on responses.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c114dcb4abce26925fa99904969d01935e9a68d">[5c114dcb4a]</a> Michael Kuron -- manager: AOC-S support for AOCMessage</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fee9012fe16cb87ccdf23ea0abdc1885e811ae4b">[fee9012fe1]</a> Michael Kuron -- res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip</li>
</ul><br><h4>Category: Functions/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29432">ASTERISK-29432</a>: New function to allow access to any channel<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ed4518328d1cf3f7903233495eeae4c76fd6a46">[1ed4518328]</a> Naveen Albert -- func_export: Add EXPORT function</li>
</ul><br><h4>Category: Functions/func_strings</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30222">ASTERISK-30222</a>: func_strings: Add trim functions<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab1dbfef754a321f7ead2163bd8f07e7eec4d30b">[ab1dbfef75]</a> Naveen Albert -- func_strings: Add trim functions.</li>
</ul><br><h4>Category: Resources/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30254">ASTERISK-30254</a>: res_tonedetect: Add audible ringback detection to TONE_DETECT<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0e7f3573012507c535a3728a94e36c022077313">[e0e7f35730]</a> Naveen Albert -- res_tonedetect: Add ringback support to TONE_DETECT.</li>
</ul><br><h4>Category: Resources/res_hep</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30322">ASTERISK-30322</a>: res_hep: Add capture agent name support<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=531eacd6c97c2c44e88b0e26c9d5c2acc4d7f463">[531eacd6c9]</a> Naveen Albert -- res_hep: Add support for named capture agents.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30032">ASTERISK-30032</a>: Support of mediasec SIP headers and SDP attributes<br/>Reported by: Maximilian Fridrich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d2e1401235da1bfca36a3b8febd0f4198737dc7">[0d2e140123]</a> Maximilian Fridrich -- res_pjsip: Add mediasec capabilities.</li>
</ul><br><h4>Category: Resources/res_pjsip_logger</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30146">ASTERISK-30146</a>: res_pjsip_logger: Add method-based log filtering<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c32b39d123e0c372ae9bd33fc3684baaa54bba80">[c32b39d123]</a> Naveen Albert -- res_pjsip_logger: Add method-based logging option.</li>
</ul><br><h4>Category: Resources/res_pjsip_notify</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30263">ASTERISK-30263</a>: res_pjsip_notify: Allow using pjsip_notify.conf from AMI<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40b52322e554fbf3cfa06dbe92732ffaaa188081">[40b52322e5]</a> Naveen Albert -- res_pjsip_notify: Add option support for AMI.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30150">ASTERISK-30150</a>: res_pjsip_session: Add support for custom parameters<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=406143ae61f7d10c5bef3891e698d9cfb43d4d0a">[406143ae61]</a> Naveen Albert -- res_pjsip_header_funcs: Add custom parameter support.</li>
</ul><br><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Resources/res_geolocation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30346">ASTERISK-30346</a>: Fix NULL dereferencing issue in Geolocation<br/>Reported by: Alexandre Fournier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=01b3962201edaa437cc67ff9fe0d80d06e958258">[01b3962201]</a> Alexandre Fournier -- res_geoloc: fix NULL pointer dereference bug</li>
</ul><br><h4>Category: Resources/res_stasis_snoop</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30252">ASTERISK-30252</a>: Unidirectional snoop on resampled channel causes garbled audio<br/>Reported by: Michael Bradeen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=907d7e7d7d2625a065a718f313d9c523e2a0ca55">[907d7e7d7d]</a> Mike Bradeen -- audiohook: add directional awareness</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=4a9f5e7ae6c4ac1e9de171c8a7fa54cd5f58a45b">4a9f5e7ae6</a></td><td>Asterisk Development Team</td><td>Update for 20.1.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aecc5448f14e6f89fb87f4b46c4808a68ec0b2ff">aecc5448f1</a></td><td>Asterisk Development Team</td><td>Update for 20.1.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fefc236e7c606fd61f662a142cc3c9d5e5133494">fefc236e7c</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 20.1.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a500b325a368d063f23dabd957b46633f0d2ade">2a500b325a</a></td><td>George Joseph</td><td>res_geolocation: Update wiki documentation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f80830ced975337a2cfcf384acfbcead72d94a2">7f80830ced</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 20.0.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=278c5726ca9d3f3df5b2c6c22e2fbcddcd95f854">278c5726ca</a></td><td>Jaco Kroon</td><td>manager: be more aggressive about purging http sessions.</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>asterisk-20.0.0-summary.html | 1836 ----
asterisk-20.0.0-summary.txt | 3804 ----------
b/.version | 2
b/CHANGES | 146
b/ChangeLog | 1188 +++
b/Makefile.rules | 4
b/UPGRADE.txt | 13
b/addons/ooh323c/src/ooq931.c | 15
b/apps/app_amd.c | 51
b/apps/app_bridgewait.c | 9
b/apps/app_if.c | 383 +
b/apps/app_meetme.c | 24
b/apps/app_mixmonitor.c | 66
b/apps/app_stack.c | 48
b/apps/app_voicemail.c | 31
b/asterisk-20.1.0-rc2-summary.html | 11
b/asterisk-20.1.0-rc2-summary.txt | 82
b/channels/chan_dahdi.c | 9
b/channels/chan_rtp.c | 2
b/channels/sig_analog.c | 28
b/configs/samples/amd.conf.sample | 7
b/configs/samples/asterisk.conf.sample | 11
b/configs/samples/cdr.conf.sample | 11
b/configs/samples/chan_dahdi.conf.sample | 9
b/configs/samples/geolocation.conf.sample | 2
b/configs/samples/hep.conf.sample | 3
b/configs/samples/pjsip.conf.sample | 12
b/configs/samples/voicemail.conf.sample | 2
b/contrib/ast-db-manage/config/versions/417c0247fd7e_add_security_negotiation_and_security_.py | 49
b/contrib/ast-db-manage/config/versions/539f68bede2c_add_peer_supported_to_100rel.py | 57
b/contrib/ast-db-manage/config/versions/5a2247c957d2_add_aoc_option.py | 38
b/contrib/ast-db-manage/config/versions/ccf795ee535f_all_codecs_on_empty_reinvite.py | 37
b/contrib/realtime/mysql/mysql_config.sql | 30
b/contrib/realtime/postgresql/postgresql_config.sql | 36
b/funcs/func_export.c | 107
b/funcs/func_json.c | 4
b/funcs/func_logic.c | 3
b/funcs/func_presencestate.c | 4
b/funcs/func_sayfiles.c | 9
b/funcs/func_scramble.c | 1
b/funcs/func_strings.c | 186
b/include/asterisk/audiohook.h | 9
b/include/asterisk/cdr.h | 2
b/include/asterisk/manager.h | 12
b/include/asterisk/res_pjsip.h | 239
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/rtp_engine.h | 12
b/include/asterisk/test.h | 10
b/main/audiohook.c | 22
b/main/cdr.c | 86
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/db.c | 4
b/main/dial.c | 9
b/main/features.c | 28
b/main/file.c | 1
b/main/manager.c | 521 +
b/main/options.c | 1
b/main/pbx_builtins.c | 17
b/main/say.c | 6
b/main/tcptls.c | 2
b/main/test.c | 9
b/main/xmldoc.c | 124
b/res/ari/resource_channels.c | 2
b/res/prometheus/bridges.c | 12
b/res/res_adsi.c | 8
b/res/res_agi.c | 15
b/res/res_crypto.c | 75
b/res/res_geolocation/geoloc_datastore.c | 4
b/res/res_geolocation/wiki/AsteriskImplementation.md | 251
b/res/res_geolocation/wiki/GML.md | 38
b/res/res_geolocation/wiki/Geolocation.md | 2
b/res/res_geolocation/wiki/README.txt | 31
b/res/res_geolocation/wiki/URI.md | 5
b/res/res_hep.c | 22
b/res/res_pjsip.c | 47
b/res/res_pjsip/config_global.c | 21
b/res/res_pjsip/pjsip_config.xml | 60
b/res/res_pjsip/pjsip_configuration.c | 45
b/res/res_pjsip/pjsip_options.c | 5
b/res/res_pjsip/pjsip_transport_events.c | 214
b/res/res_pjsip/security_agreements.c | 340
b/res/res_pjsip_aoc.c | 698 +
b/res/res_pjsip_exten_state.c | 13
b/res/res_pjsip_header_funcs.c | 268
b/res/res_pjsip_logger.c | 165
b/res/res_pjsip_mwi.c | 13
b/res/res_pjsip_notify.c | 165
b/res/res_pjsip_outbound_authenticator_digest.c | 4
b/res/res_pjsip_outbound_registration.c | 253
b/res/res_pjsip_pubsub.c | 25
b/res/res_pjsip_rfc3329.c | 150
b/res/res_pjsip_sdp_rtp.c | 6
b/res/res_pjsip_session.c | 118
b/res/res_pjsip_transport_websocket.c | 1
b/res/res_stasis_snoop.c | 22
b/res/res_tonedetect.c | 25
b/tests/CI/runUnittests.sh | 8
b/tests/test_crypto.c | 19
b/tests/test_json.c | 28
b/tests/test_mwi.c | 4
b/tests/test_stasis.c | 2
b/tests/test_stasis_state.c | 8
b/third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch | 300
104 files changed, 6855 insertions(+), 6136 deletions(-)</pre><br></html>

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Release Summary
asterisk-20.1.0
Date: 2023-01-12
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Open Issues
5. Other Changes
6. 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-20.0.0.
----------------------------------------------------------------------
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
39 Naveen Albert 1 inspecting SIP traces. 37 N A
7 George Joseph 7 George Joseph
5 Philip Prindeville 4 Philip Prindeville
4 Asterisk Development Team 3 Matt Jordan
4 Mike Bradeen 3 Maximilian Fridrich
3 Maximilian Fridrich 2 Michael Bradeen
3 Joshua C. Colp 2 Marcel Wagner
2 Jaco Kroon 2 Joshua C. Colp
2 Michael Kuron 1 Igor Yeroshev
2 Marcel Wagner 1 David Uczen
1 Birger Harzenetter 1 Alexandre Fournier
(license 5870) 1 Igor Goncharovsky
1 Holger Hans Peter 1 Frederic LE FOLL
Freyther 1 shawty
1 Igor Goncharovsky 1 David Woolley
1 Frederic LE FOLL 1 nappsoft
1 Alexandre Fournier 1 Jesse Ross
1 Ben Ford 1 Joeran Vinzens
1 Henning Westerholt 1 Henning Westerholt
1 Benjamin Keith Ford
1 Saken
----------------------------------------------------------------------
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.
Security
Category: Addons/chan_ooh323
ASTERISK-30103: chan_ooh323 Vulnerability in calling/called party IE
Reported by: Michael Bradeen
* [eb1d7ab53c] Mike Bradeen -- ooh323c: not checking for IE minimum
length
Category: Core/ManagerInterface
ASTERISK-30176: manager: GetConfig can read files outside of Asterisk
Reported by: shawty
* [81f10e847e] Mike Bradeen -- manager: prevent file access outside of
config dir
Category: pjproject/pjsip
ASTERISK-30338: pjproject: Backport security fixes from 2.13
Reported by: Benjamin Keith Ford
* [d476994768] Ben Ford -- pjproject: 2.13 security fixes
Improvement
Category: Applications/app_mixmonitor
ASTERISK-30286: app_mixmonitor: Add option to use real Caller ID for
Caller ID
Reported by: N A
* [691178c48e] Naveen Albert -- app_mixmonitor: Add option to use real
Caller ID for voicemail.
Category: Core/Bridging
ASTERISK-30223: features: add no-answer option to Bridge application
Reported by: N A
* [a5ec60e6c6] Naveen Albert -- features: Add no answer option to
Bridge.
Category: Core/General
ASTERISK-30290: file.c: Don't emit warnings on winks.
Reported by: N A
* [8142b313c3] Naveen Albert -- file.c: Don't emit warnings on winks.
Category: Core/RTP
ASTERISK-30327: rtp_engine.h: Remove obsolete example usage
Reported by: N A
* [a4bcdce1db] Naveen Albert -- rtp_engine.h: Update examples using
ast_format_set.
Category: Documentation
ASTERISK-30328: Typo in from_domain description on res_pjsip configuration
documentation
Reported by: Marcel Wagner
* [58534b309f] Marcel Wagner -- res_pjsip: Fix typo in from_domain
documentation
ASTERISK-30316: res_pjsip: Documentation should point out default if
contact_user is not being set for outbound registrations
Reported by: Marcel Wagner
* [b83af13f65] Marcel Wagner -- res_pjsip: Update contact_user to point
out default
ASTERISK-30289: xmldoc: Allow XML docs to be reloaded
Reported by: N A
* [52c7d3ed07] Naveen Albert -- xmldoc: Allow XML docs to be reloaded.
Category: PBX/General
ASTERISK-30308: pbx_builtins: Allow Answer to return immediately
Reported by: N A
* [c7df5ee7c1] Naveen Albert -- pbx_builtins: Allow Answer to return
immediately.
Category: Resources/res_pjsip
ASTERISK-30158: PJSIP: Add new 100rel option "peer_supported"
Reported by: Maximilian Fridrich
* [5bbad0d27c] Maximilian Fridrich -- res_pjsip: Add 100rel option
"peer_supported".
Category: Resources/res_pjsip_geolocation
ASTERISK-30241: res_pjsip_gelocation: Downgrade some NOTICE scope trace
debugs to DEBUG level
Reported by: N A
* [8afb313a43] Naveen Albert -- res_pjsip_geolocation: Change some
notices to debugs.
Category: Tests/General
ASTERISK-30295: test_json: Remove duplicated static function
Reported by: N A
* [1e77b8c473] Naveen Albert -- test_json: Remove duplicated static
function.
Bug
Category: Applications/app_if
ASTERISK-30349: app_if: Format truncation error
Reported by: George Joseph
* [2f9cdfbc50] Naveen Albert -- app_if: Fix format truncation errors.
Category: Applications/app_meetme
ASTERISK-29998: sla: deadlock when calling SLAStation application
Reported by: N A
* [2efa290d3c] Naveen Albert -- sla: Prevent deadlock and crash due to
autoservicing.
Category: Applications/app_saynumber
ASTERISK-30248: ast_get_digit_str adds bogus initial delimiter if first
character not to be spoken
Reported by: David Woolley
* [407216a0a5] Naveen Albert -- say: Don't prepend ampersand
erroneously.
Category: Applications/app_stack
ASTERISK-30076: app_stack: Incorrect exit location in predial handlers
logged
Reported by: N A
* [dfe2f38642] Naveen Albert -- app_stack: Print proper exit location
for PBXless channels.
Category: Applications/app_voicemail
ASTERISK-30283: app_voicemail: Fix msg_create_from_file not sending email
to user
Reported by: N A
* [b9c031c1f8] Naveen Albert -- app_voicemail: Fix missing email in
msg_create_from_file.
ASTERISK-30248: ast_get_digit_str adds bogus initial delimiter if first
character not to be spoken
Reported by: David Woolley
* [407216a0a5] Naveen Albert -- say: Don't prepend ampersand
erroneously.
Category: Channels/chan_dahdi
ASTERISK-30336: sig_analog: Fix no timeout duration
Reported by: N A
* [b90e57758b] Naveen Albert -- sig_analog: Fix no timeout duration.
ASTERISK-30274: chan_dahdi: Unavailable channels are BUSY
Reported by: N A
* [180ca32565] Naveen Albert -- chan_dahdi: Fix unavailable channels
returning busy.
ASTERISK-30256: chan_dahdi: Fix format truncation warnings
Reported by: N A
* [98fc05f13b] Naveen Albert -- chan_dahdi: Resolve format truncation
warning.
Category: Channels/chan_local
ASTERISK-30239: Prometheus plugin crashes Asterisk when using local
channel
Reported by: Joeran Vinzens
* [62881c668b] Holger Hans Peter Freyther -- res_prometheus: Do not
crash on invisible bridges
Category: Channels/chan_pjsip
ASTERISK-30184: res_pjsip_session: re-INVITE after answering results in
wrong stream direction of first call leg
Reported by: Maximilian Fridrich
* [60b81eabe0] Maximilian Fridrich -- core & res_pjsip: Improve topology
change handling.
ASTERISK-30193: chan_pjsip should return all codecs on a re-INVITE without
SDP
Reported by: Henning Westerholt
* [12445040d3] Henning Westerholt -- res_pjsip: return all codecs on a
re-INVITE without SDP
Category: Channels/chan_rtp
ASTERISK-30281: chan_rtp: Local address being used before being set
Reported by: George Joseph
* [f723b465e5] George Joseph -- chan_rtp: Make usage of
ast_rtp_instance_get_local_address clearer
Category: Core/AstDB
ASTERISK-30245: db: ListItems is incorrect
Reported by: N A
* [7335b0cffe] Birger Harzenetter -- db: Fix incorrect DB tree count for
AMI.
Category: Core/BuildSystem
ASTERISK-30321: Build: Embedded blobs have executable stacks
Reported by: George Joseph
* [ce2153fc5a] Jaco Kroon -- Build system: Avoid executable stack.
Category: Core/Dial
ASTERISK-30258: Dialing API: Cancel a running async thread, does not
always cancel all calls
Reported by: Frederic LE FOLL
* [50a4495799] Frederic LE FOLL -- Dialing API: Cancel a running async
thread, may not cancel all calls
Category: Core/General
ASTERISK-30278: tcptls: Abort occurs if SSL error is logged if
MALLOC_DEBUG is enabled
Reported by: N A
* [afd86b47c1] Naveen Albert -- tcptls: Prevent crash when freeing
OpenSSL errors.
Category: Core/HTTP
ASTERISK-30278: tcptls: Abort occurs if SSL error is logged if
MALLOC_DEBUG is enabled
Reported by: N A
* [afd86b47c1] Naveen Albert -- tcptls: Prevent crash when freeing
OpenSSL errors.
Category: Core/ManagerInterface
ASTERISK-30285: manager.c: Remove outdated documentation
Reported by: N A
* [49cfdbbdff] Naveen Albert -- manager: Update ModuleCheck
documentation.
Category: Functions/General
ASTERISK-30293: Memory leak in JSON_DECODE
Reported by: David Uczen
* [002afc3f2a] Naveen Albert -- func_json: Fix memory leak.
ASTERISK-30220: func_scramble: Fix segfault due to null pointer deref
Reported by: N A
* [8aae0b9f08] Naveen Albert -- func_scramble: Fix null pointer
dereference.
Category: Functions/func_logic
ASTERISK-30243: func_logic: IF function complains if both branches are
empty
Reported by: N A
* [407167cc28] Naveen Albert -- func_logic: Don't emit warning if both
IF branches are empty.
Category: Functions/func_presencestate
ASTERISK-30311: func_presencestate: Fix invalid memory access.
Reported by: N A
* [83eb113e0f] Naveen Albert -- func_presencestate: Fix invalid memory
access.
Category: Resources/General
ASTERISK-30239: Prometheus plugin crashes Asterisk when using local
channel
Reported by: Joeran Vinzens
* [62881c668b] Holger Hans Peter Freyther -- res_prometheus: Do not
crash on invisible bridges
ASTERISK-30237: res_prometheus: Crash when scraping bridges
Reported by: Igor Yeroshev
* [62881c668b] Holger Hans Peter Freyther -- res_prometheus: Do not
crash on invisible bridges
Category: Resources/res_adsi
ASTERISK-29793: adsi: CAS is malformed
Reported by: N A
* [80e6205bb0] Naveen Albert -- res_adsi: Fix major regression caused by
media format rearchitecture.
Category: Resources/res_agi
ASTERISK-30314: res_agi: RECORD FILE doesn't respect "transmit_silence"
asterisk.conf option
Reported by: Joshua C. Colp
* [61922d2934] Joshua C. Colp -- res_agi: Respect "transmit_silence"
option for "RECORD FILE".
Category: Resources/res_ari
ASTERISK-30344: ari: Memory leak in create when specifying JSON
Reported by: Saken
* [564349ff5d] Joshua C. Colp -- ari: Destroy body variables in channel
create.
Category: Resources/res_ari_channels
ASTERISK-30237: res_prometheus: Crash when scraping bridges
Reported by: Igor Yeroshev
* [62881c668b] Holger Hans Peter Freyther -- res_prometheus: Do not
crash on invisible bridges
Category: Resources/res_crypto
ASTERISK-30213: Make crypto_load() reentrant and handle symlinks correctly
Reported by: Philip Prindeville
* [d0bea5a725] Philip Prindeville -- res_crypto: handle unsafe private
key files
* [ef74ecacc7] Philip Prindeville -- res_crypto: don't modify fname in
try_load_key()
* [5e2485b5c0] Philip Prindeville -- res_crypto: use
ast_file_read_dirs() to iterate
ASTERISK-30235: res_crypto and tests: Memory issues and and uninitialized
variable error
Reported by: George Joseph
* [e25b690d10] George Joseph -- res_crypto: Memory issues and
uninitialized variable errors
ASTERISK-30226: REGRESSION: res_crypto complains about the stir_shaken
directory in /var/lib/asterisk/keys
Reported by: George Joseph
* [026dc08529] Philip Prindeville -- res_crypto: don't complain about
directories
Category: Resources/res_geolocation
ASTERISK-30234: res_geolocation: ...may be used uninitialized error in
geoloc_config.c
Reported by: George Joseph
* [e33f2dcc0f] George Joseph -- res_geolocation: Fix issues exposed by
compiling with -O2
Category: Resources/res_pjsip
ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they
shouldn't be
Reported by: Joshua C. Colp
* [ff9ab38573] George Joseph -- res_pjsip_transport_websocket: Add
remote port to transport
ASTERISK-30184: res_pjsip_session: re-INVITE after answering results in
wrong stream direction of first call leg
Reported by: Maximilian Fridrich
* [60b81eabe0] Maximilian Fridrich -- core & res_pjsip: Improve topology
change handling.
Category: Resources/res_pjsip_exten_state
ASTERISK-30264: res_pjsip: Subscription handlers do not get cleanly
unregistered, causing crash
Reported by: N A
* [9258d8212a] Naveen Albert -- res_pjsip_pubsub: Prevent removing
subscriptions.
Category: Resources/res_pjsip_mwi
ASTERISK-30264: res_pjsip: Subscription handlers do not get cleanly
unregistered, causing crash
Reported by: N A
* [9258d8212a] Naveen Albert -- res_pjsip_pubsub: Prevent removing
subscriptions.
Category: Resources/res_pjsip_outbound_registration
ASTERISK-30217: Registration do not allow multiple proxies
Reported by: Igor Goncharovsky
* [096529d33f] Igor Goncharovsky -- res_pjsip_outbound_registration:
Allow to use multiple proxies for registration
Category: Resources/res_pjsip_pubsub
ASTERISK-30244: res_pjsip_pubsub: Occasional crash when TCP/TLS connection
terminated and subscription persistence is removed
Reported by: nappsoft
* [7684c9e907] George Joseph -- pjsip_transport_events: Fix possible use
after free on transport
Category: Resources/res_pjsip_session
ASTERISK-30265: res_pjsip_session: Fix missing PLAR support on INVITEs
Reported by: N A
* [0d6003fa9a] Naveen Albert -- res_pjsip_session.c: Map empty
extensions in INVITEs to s.
ASTERISK-30184: res_pjsip_session: re-INVITE after answering results in
wrong stream direction of first call leg
Reported by: Maximilian Fridrich
* [60b81eabe0] Maximilian Fridrich -- core & res_pjsip: Improve topology
change handling.
Category: Tests/General
ASTERISK-30282: CI: Coredump output isn't saved when running unittests
Reported by: George Joseph
* [0c1c623dee] George Joseph -- runUnittests.sh: Save coredumps to
proper directory
ASTERISK-30232: Initialize stack-based ast_test_capture structures
correctly
Reported by: Philip Prindeville
* [3e7ce90f9c] Philip Prindeville -- test: initialize capture structure
before freeing
Category: Tests/Portability
ASTERISK-30273: test_mwi: compilation fails on 32-bit Debian
Reported by: N A
* [ca8900b0f6] Naveen Albert -- tests: Fix compilation errors on 32-bit.
Category: pjproject/pjsip
ASTERISK-28689: res_pjsip: Crash when locking group lock when sending
stateful response
Reported by: Jesse Ross
* [50e2921a48] Mike Bradeen -- res_pjsip: prevent crash on websocket
disconnect
New Feature
Category: Applications/NewFeature
ASTERISK-29497: Add conditional branch applications
Reported by: N A
* [b365ea8601] Naveen Albert -- app_if: Adds conditional branch
applications
Category: Applications/app_amd
ASTERISK-30179: app_amd: Allow audio to be played while AMD is running
Reported by: N A
* [8c791f9a65] Naveen Albert -- app_amd: Add option to play audio during
AMD.
Category: Applications/app_bridgewait
ASTERISK-30216: app_bridgewait: Add option for BridgeWait to not answer
Reported by: N A
* [1e29607b5c] Naveen Albert -- app_bridgewait: Add option to not answer
channel.
Category: Applications/app_mixmonitor
ASTERISK-30284: app_mixmonitor: Add option to delete recording file when
done
Reported by: N A
* [6e59b01e1a] Naveen Albert -- app_mixmonitor: Add option to delete
files on exit.
Category: CDR/General
ASTERISK-30091: cdr: Allow CDRs to ignore call state changes
Reported by: N A
* [b331caca30] Naveen Albert -- cdr: Allow bridging and dial state
changes to be ignored.
Category: Channels/chan_dahdi
ASTERISK-30305: chan_dahdi: Allow FXO channels to start immediately
Reported by: N A
* [5ede4e217a] Naveen Albert -- chan_dahdi: Allow FXO channels to start
immediately.
Category: Channels/chan_pjsip
ASTERISK-21502: New SIP Channel Driver - add Advice of Charge support
Reported by: Matt Jordan
* [b6855755ce] Joshua C. Colp -- res_pjsip_aoc: Don't assume a body
exists on responses.
* [5c114dcb4a] Michael Kuron -- manager: AOC-S support for AOCMessage
* [fee9012fe1] Michael Kuron -- res_pjsip_aoc: New module for sending
advice-of-charge with chan_pjsip
Category: Functions/NewFeature
ASTERISK-29432: New function to allow access to any channel
Reported by: N A
* [1ed4518328] Naveen Albert -- func_export: Add EXPORT function
Category: Functions/func_strings
ASTERISK-30222: func_strings: Add trim functions
Reported by: N A
* [ab1dbfef75] Naveen Albert -- func_strings: Add trim functions.
Category: Resources/General
ASTERISK-30254: res_tonedetect: Add audible ringback detection to
TONE_DETECT
Reported by: N A
* [e0e7f35730] Naveen Albert -- res_tonedetect: Add ringback support to
TONE_DETECT.
Category: Resources/res_hep
ASTERISK-30322: res_hep: Add capture agent name support
Reported by: N A
* [531eacd6c9] Naveen Albert -- res_hep: Add support for named capture
agents.
Category: Resources/res_pjsip
ASTERISK-30032: Support of mediasec SIP headers and SDP attributes
Reported by: Maximilian Fridrich
* [0d2e140123] Maximilian Fridrich -- res_pjsip: Add mediasec
capabilities.
Category: Resources/res_pjsip_logger
ASTERISK-30146: res_pjsip_logger: Add method-based log filtering
Reported by: N A
* [c32b39d123] Naveen Albert -- res_pjsip_logger: Add method-based
logging option.
Category: Resources/res_pjsip_notify
ASTERISK-30263: res_pjsip_notify: Allow using pjsip_notify.conf from AMI
Reported by: N A
* [40b52322e5] Naveen Albert -- res_pjsip_notify: Add option support for
AMI.
Category: Resources/res_pjsip_session
ASTERISK-30150: res_pjsip_session: Add support for custom parameters
Reported by: N A
* [406143ae61] Naveen Albert -- res_pjsip_header_funcs: Add custom
parameter support.
----------------------------------------------------------------------
Open Issues
[Back to Top]
This is a list of all open issues from the issue tracker that were
referenced by changes that went into this release.
Bug
Category: Resources/res_geolocation
ASTERISK-30346: Fix NULL dereferencing issue in Geolocation
Reported by: Alexandre Fournier
* [01b3962201] Alexandre Fournier -- res_geoloc: fix NULL pointer
dereference bug
Category: Resources/res_stasis_snoop
ASTERISK-30252: Unidirectional snoop on resampled channel causes garbled
audio
Reported by: Michael Bradeen
* [907d7e7d7d] Mike Bradeen -- audiohook: add directional awareness
----------------------------------------------------------------------
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 |
|------------+----------------------+------------------------------------|
| 4a9f5e7ae6 | Asterisk Development | Update for 20.1.0-rc2 |
| | Team | |
|------------+----------------------+------------------------------------|
| aecc5448f1 | Asterisk Development | Update for 20.1.0-rc1 |
| | Team | |
|------------+----------------------+------------------------------------|
| fefc236e7c | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 20.1.0 |
|------------+----------------------+------------------------------------|
| 2a500b325a | George Joseph | res_geolocation: Update wiki |
| | | documentation |
|------------+----------------------+------------------------------------|
| 7f80830ced | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 20.0.0 |
|------------+----------------------+------------------------------------|
| 278c5726ca | Jaco Kroon | manager: be more aggressive about |
| | | purging http sessions. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
asterisk-20.0.0-summary.html | 1836 ----
asterisk-20.0.0-summary.txt | 3804 ----------
b/.version | 2
b/CHANGES | 146
b/ChangeLog | 1188 +++
b/Makefile.rules | 4
b/UPGRADE.txt | 13
b/addons/ooh323c/src/ooq931.c | 15
b/apps/app_amd.c | 51
b/apps/app_bridgewait.c | 9
b/apps/app_if.c | 383 +
b/apps/app_meetme.c | 24
b/apps/app_mixmonitor.c | 66
b/apps/app_stack.c | 48
b/apps/app_voicemail.c | 31
b/asterisk-20.1.0-rc2-summary.html | 11
b/asterisk-20.1.0-rc2-summary.txt | 82
b/channels/chan_dahdi.c | 9
b/channels/chan_rtp.c | 2
b/channels/sig_analog.c | 28
b/configs/samples/amd.conf.sample | 7
b/configs/samples/asterisk.conf.sample | 11
b/configs/samples/cdr.conf.sample | 11
b/configs/samples/chan_dahdi.conf.sample | 9
b/configs/samples/geolocation.conf.sample | 2
b/configs/samples/hep.conf.sample | 3
b/configs/samples/pjsip.conf.sample | 12
b/configs/samples/voicemail.conf.sample | 2
b/contrib/ast-db-manage/config/versions/417c0247fd7e_add_security_negotiation_and_security_.py | 49
b/contrib/ast-db-manage/config/versions/539f68bede2c_add_peer_supported_to_100rel.py | 57
b/contrib/ast-db-manage/config/versions/5a2247c957d2_add_aoc_option.py | 38
b/contrib/ast-db-manage/config/versions/ccf795ee535f_all_codecs_on_empty_reinvite.py | 37
b/contrib/realtime/mysql/mysql_config.sql | 30
b/contrib/realtime/postgresql/postgresql_config.sql | 36
b/funcs/func_export.c | 107
b/funcs/func_json.c | 4
b/funcs/func_logic.c | 3
b/funcs/func_presencestate.c | 4
b/funcs/func_sayfiles.c | 9
b/funcs/func_scramble.c | 1
b/funcs/func_strings.c | 186
b/include/asterisk/audiohook.h | 9
b/include/asterisk/cdr.h | 2
b/include/asterisk/manager.h | 12
b/include/asterisk/res_pjsip.h | 239
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/rtp_engine.h | 12
b/include/asterisk/test.h | 10
b/main/audiohook.c | 22
b/main/cdr.c | 86
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/db.c | 4
b/main/dial.c | 9
b/main/features.c | 28
b/main/file.c | 1
b/main/manager.c | 521 +
b/main/options.c | 1
b/main/pbx_builtins.c | 17
b/main/say.c | 6
b/main/tcptls.c | 2
b/main/test.c | 9
b/main/xmldoc.c | 124
b/res/ari/resource_channels.c | 2
b/res/prometheus/bridges.c | 12
b/res/res_adsi.c | 8
b/res/res_agi.c | 15
b/res/res_crypto.c | 75
b/res/res_geolocation/geoloc_datastore.c | 4
b/res/res_geolocation/wiki/AsteriskImplementation.md | 251
b/res/res_geolocation/wiki/GML.md | 38
b/res/res_geolocation/wiki/Geolocation.md | 2
b/res/res_geolocation/wiki/README.txt | 31
b/res/res_geolocation/wiki/URI.md | 5
b/res/res_hep.c | 22
b/res/res_pjsip.c | 47
b/res/res_pjsip/config_global.c | 21
b/res/res_pjsip/pjsip_config.xml | 60
b/res/res_pjsip/pjsip_configuration.c | 45
b/res/res_pjsip/pjsip_options.c | 5
b/res/res_pjsip/pjsip_transport_events.c | 214
b/res/res_pjsip/security_agreements.c | 340
b/res/res_pjsip_aoc.c | 698 +
b/res/res_pjsip_exten_state.c | 13
b/res/res_pjsip_header_funcs.c | 268
b/res/res_pjsip_logger.c | 165
b/res/res_pjsip_mwi.c | 13
b/res/res_pjsip_notify.c | 165
b/res/res_pjsip_outbound_authenticator_digest.c | 4
b/res/res_pjsip_outbound_registration.c | 253
b/res/res_pjsip_pubsub.c | 25
b/res/res_pjsip_rfc3329.c | 150
b/res/res_pjsip_sdp_rtp.c | 6
b/res/res_pjsip_session.c | 118
b/res/res_pjsip_transport_websocket.c | 1
b/res/res_stasis_snoop.c | 22
b/res/res_tonedetect.c | 25
b/tests/CI/runUnittests.sh | 8
b/tests/test_crypto.c | 19
b/tests/test_json.c | 28
b/tests/test_mwi.c | 4
b/tests/test_stasis.c | 2
b/tests/test_stasis_state.c | 8
b/third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch | 300
104 files changed, 6855 insertions(+), 6136 deletions(-)