kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2026-01-01 04:05:02 +00:00
Code Issues Packages Projects Releases Wiki Activity
Files
39dc1ac0c7a5c90d7613de644b4e3eb16fed121e
asterisk/asterisk-16.9.0-summary.html
Asterisk Development Team 39dc1ac0c7 Update for 16.9.0
2020-03-12 09:44:05 -05:00

160 lines
28 KiB
HTML
Raw Blame History

<!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-16.9.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.9.0</h3><h3 align="center">Date: 2020-03-12</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="#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-16.8.0.</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%">11 Sean Bright <sean.bright@gmail.com><br/>7 Joshua C. Colp <jcolp@sangoma.com><br/>5 Kevin Harwell <kharwell@digium.com><br/>5 Walter Doekes <walter+github@wjd.nu><br/>3 George Joseph <gjoseph@digium.com><br/>2 Torrey Searle <torrey@voxbone.com><br/>2 Asterisk Development Team <asteriskteam@digium.com><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><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/>3 Joshua C. Colp <jcolp@digium.com><br/>2 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Paul Brooks<br/>1 Martin Zeh<br/>1 Sébastien Duthil <sduthil@wazo.community><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Kevin Harwell <kharwell@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 EDV O-TON<br/>1 Peter Sokolov<br/>1 George Joseph <gjoseph@digium.com><br/>1 Dmitriy Serov <serov.d.p@gmail.com><br/>1 Dmitriy Serov<br/>1 Alex <alex@alex-at.ru><br/>1 Sébastien Duthil<br/>1 Torrey Searle <tsearle@gmail.com><br/>1 lvl <digium@lvlconsultancy.nl><br/>1 Benjamin Keith Ford <bford@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: 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=f529d56f760c0c16bd7ec53428b53749777c8234">[f529d56f76]</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=fc1d93cf97e6d059584b1feb5d075d22ead31669">[fc1d93cf97]</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=34ce90a9e7db378bb3a7748de8ae9b23657c6a1a">[34ce90a9e7]</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=ad9a40a539d9b3462049f41ee042de10125dce98">[ad9a40a539]</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=d8d989b9bf8d6d74730fea2e8938eafd19fc582b">[d8d989b9bf]</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=fc1d93cf97e6d059584b1feb5d075d22ead31669">[fc1d93cf97]</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=dc9875815c441bcb78370cbf0d331ec46e7abb1d">[dc9875815c]</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=1e94cfdf4c00db6b96c6740d4ef66a02acb3e210">[1e94cfdf4c]</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=1532da300894dccfdc5bb5b5dfeed9c55e42ca25">[1532da3008]</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=b20a215ea703a4d689c6cfa4cb974ee196fb5e1c">[b20a215ea7]</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=2a9a1789e78ac6f34a17b3d9f83896d94ea74c39">[2a9a1789e7]</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=4bbf24d2ff35e27f800766459eff010b66688274">[4bbf24d2ff]</a> Sean Bright -- res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28746">ASTERISK-28746</a>: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7eab841093d3a91a38f9836b1c54199821a35429">[7eab841093]</a> George Joseph -- res_pjsip_outbound_registration: Fix SRV failover on timeout</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=4d32f5747c2db25d63005bbb94ee9d27e09059c2">[4d32f5747c]</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=9ecb6515642fa890191dc28f0a50dbffc6f946d4">[9ecb651564]</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=afcc838796552388055d9cfa44abfdc2b63e6b3f">[afcc838796]</a> Torrey Searle -- res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28730">ASTERISK-28730</a>: res_pjsip_session: Fix out of order session refreshes<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b438d1d9adc8c5200214e41876e9abc02c5b5288">[b438d1d9ad]</a> Joshua C. Colp -- res_pjsip_session: Fix off-nominal session refreshes.</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28764">ASTERISK-28764</a>: res_rtp_asterisk: Improve NACK support and seqno handling<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f295af447db656d14218f7b61c4bd7bd78d0b194">[f295af447d]</a> Joshua C. Colp -- res_rtp_asterisk: Improve video performance in certain networks.</li>
</ul><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=411d3a2f75bc42614f986e9c1c171e258f44730f">[411d3a2f75]</a> Ben Ford -- RTP/ICE: Send on first valid pair.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28742">ASTERISK-28742</a>: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb783b0c1108024571df33fd0d5248ca5420c43f">[bb783b0c11]</a> Kevin Harwell -- res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup</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=e103339f02f0445b8c77b1c3c6f7d1c80e37f675">[e103339f02]</a> Kevin Harwell -- stasis/app: don't lock an app before a call to send</li>
</ul><br><h4>Category: Resources/res_stasis_playback</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28713">ASTERISK-28713</a>: res_stasis_playback: Error building JSON<br/>Reported by: Sébastien Duthil<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a6b09e2e8e26736f1836d2e66fe464592b16a4f">[2a6b09e2e8]</a> Sean Bright -- res_stasis_playback: Prevent media_index from going out of bounds</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=06e8d5ad8e4728a716bf357c8d7f70367ae10280">[06e8d5ad8e]</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=50d91ad7070cbd1e87fbcc31185f43ad33ab7e49">[50d91ad707]</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=cd8b27dcc24d95b42263f1a826645d44029f0f8a">[cd8b27dcc2]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=423b0e68ceff0af368e83f0acc07ae7682c76751">[423b0e68ce]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=423b0e68ceff0af368e83f0acc07ae7682c76751">[423b0e68ce]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=423b0e68ceff0af368e83f0acc07ae7682c76751">[423b0e68ce]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</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=40b834ad8c5e341243f7f678efa9522b7218e5e1">[40b834ad8c]</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=c32b4c7dc0823e57b78361eff531ef3b3be514cb">[c32b4c7dc0]</a> Sean Bright -- tcptls.c: Log more informative OpenSSL errors</li>
</ul><br><h4>Category: Core/Streams</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=423b0e68ceff0af368e83f0acc07ae7682c76751">[423b0e68ce]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</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=cd8b27dcc24d95b42263f1a826645d44029f0f8a">[cd8b27dcc2]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=423b0e68ceff0af368e83f0acc07ae7682c76751">[423b0e68ce]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</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=546f93d46fa769ce0572d36b380bbf0e1cf8e9b7">546f93d46f</a></td><td>Asterisk Development Team</td><td>Update for 16.9.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1522c4467c7696147c80c2a7adf4f4f19998e6c4">1522c4467c</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.9.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad5a81aac4841db5bccc622f3e5c35c06b276e8c">ad5a81aac4</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=b12ecbe27fbc815ab6241d7f9246816b2c834015">b12ecbe27f</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=dd959bf8d02e82f02cfa181d714d61ccf8a2cc59">dd959bf8d0</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=7d1305ac21506b9536c5dc82a694321cafa4c384">7d1305ac21</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=cf1f2dfe8eeadf8c0b0d37f5ac3a0c2a4807a897">cf1f2dfe8e</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=df52f713f5dd6cc13853df326b0ad1326174e96b">df52f713f5</a></td><td>Joshua C. Colp</td><td>stasis: Use format specifier for size_t.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=65ff4d80a10600cbcc539e0e82ed1dbab2655d4d">65ff4d80a1</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=1544f749326dc6d8c5330d3a7348efea6b2bc396">1544f74932</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=35b4f1686b382bf7586350a3da8776ed4b0b19f7">35b4f1686b</a></td><td>Walter Doekes</td><td>chan_sip: Clarify in sample docs how directmediapermit/-acl should be used</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4132eec197e3c3f988e0a958d8a97fde9c1733c">f4132eec19</a></td><td>Joshua C. Colp</td><td>res_rtp_asterisk: Don't produce transport-cc if no packets.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d6574cb7c7f17afb0a463dafae81bbdaa1fa79c8">d6574cb7c7</a></td><td>George Joseph</td><td>message.c: Add option to suppress the Message channel AMI and ARI events</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-16.8.0-summary.html | 201 ---
asterisk-16.8.0-summary.txt | 620 ---------
b/.version | 2
b/CHANGES | 25
b/ChangeLog | 660 +++++++++-
b/UPGRADE.txt | 56
b/addons/res_config_mysql.c | 16
b/apps/app_mixmonitor.c | 29
b/apps/app_queue.c | 58
b/asterisk-16.9.0-rc1-summary.html | 163 ++
b/asterisk-16.9.0-rc1-summary.txt | 460 ++++++
b/bridges/bridge_native_rtp.c | 168 ++
b/bridges/bridge_simple.c | 198 +--
b/bridges/bridge_softmix.c | 246 ++-
b/channels/chan_pjsip.c | 13
b/channels/chan_sip.c | 126 +
b/channels/sip/include/sip.h | 1
b/configs/samples/asterisk.conf.sample | 5
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/CHANGES-staging/hide_messaging_ami_events | 11
b/doc/UPGRADE-staging/README.md | 7
b/funcs/func_odbc.c | 4
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/channel.h | 20
b/include/asterisk/format_cap.h | 4
b/include/asterisk/message.h | 13
b/include/asterisk/options.h | 3
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/say.h | 4
b/include/asterisk/sorcery.h | 27
b/main/asterisk.c | 1
b/main/channel.c | 19
b/main/file.c | 2
b/main/format_cap.c | 4
b/main/message.c | 27
b/main/options.c | 2
b/main/say.c | 12
b/main/sorcery.c | 46
b/main/stasis.c | 4
b/main/stream.c | 22
b/main/tcptls.c | 29
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 | 2
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_outbound_registration.c | 49
b/res/res_pjsip_refer.c | 7
b/res/res_pjsip_sdp_rtp.c | 56
b/res/res_pjsip_session.c | 107 +
b/res/res_rtp_asterisk.c | 312 +++-
b/res/res_sorcery_config.c | 1
b/res/res_stasis_playback.c | 4
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 | 8
doc/CHANGES-staging/res_fax_negotiate_both | 7
69 files changed, 2963 insertions(+), 1407 deletions(-)</pre><br></html>
Reference in New Issue View Git Blame Copy Permalink