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Asterisk Development Team
e032ae6226 Update for 17.9.0 2020-11-19 07:37:25 -05:00
Asterisk Development Team
baf66a5816 Update for 17.9.0-rc1 2020-11-12 07:19:31 -05:00
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-17.9.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.9.0</h3><h3 align="center">Date: 2020-11-19</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-17.8.0.</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%">9 Sean Bright <sean.bright@gmail.com><br/>6 George Joseph <gjoseph@digium.com><br/>3 Asterisk Development Team <asteriskteam@digium.com><br/>3 Joshua C. Colp <jcolp@sangoma.com><br/>3 Alexander Traud <pabstraud@compuserve.com><br/>2 Kevin Harwell <kharwell@sangoma.com><br/>2 Torrey Searle <tsearle@voxbone.com><br/>2 Ben Ford <bford@digium.com><br/>2 Sungtae Kim <pchero21@gmail.com><br/>1 Holger Hans Peter Freyther <holger@moiji-mobile.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Michal Hajek <michal.hajek@daktela.com><br/>1 Jean Aunis <jean.aunis@prescom.fr><br/>1 Nick French <nickfrench@gmail.com><br/>1 laszlovl <digium@lvlconsultancy.nl><br/>1 Jasper van der Neut <jasper@isotopic.nl><br/>1 Andrew Siplas <andrew@asiplas.net><br/></td><td width="33%"><td width="33%">2 Sebastian Damm <damm@sipgate.de><br/>2 George Joseph <gjoseph@digium.com><br/>2 sungtae kim <pchero21@gmail.com><br/>1 Sandro Gauci <sandro@enablesecurity.com><br/>1 Nick French <nickfrench@gmail.com><br/>1 Michal Hajek <michal.hajek@daktela.com><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 under <under@list.ru><br/>1 周家建 <zhou_0611@163.com><br/>1 Torrey Searle <tsearle@gmail.com><br/>1 Hajek Michal <michal.hajek@daktela.com><br/>1 Vieri <vieridipaola@gmail.com><br/>1 Péter Juhász <peter.juhasz@comnica.com><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 laszlovl <digium@lvlconsultancy.nl><br/>1 Thomas Frederiksen <tommer@nicesurprise.com><br/>1 Eric Smith <abkowald@gmail.com><br/>1 Brian J. Murrell <brian@interlinx.bc.ca><br/>1 Jasper van der Neut <jasper@isotopic.nl><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29057">ASTERISK-29057</a>: pjsip: Crash on call rejection during high load<br/>Reported by: Sandro Gauci<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fab1ef83b283d1e28d0ab442ccf82ab1301f9ddb">[fab1ef83b2]</a> Kevin Harwell -- AST-2020-001 - res_pjsip: Return dialog locked and referenced</li>
</ul><br><h3>Improvement</h3><h4>Category: Resources/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29056">ASTERISK-29056</a>: Increase reg_server column size for ps_contacts table realtime<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed361bc51513ca286df597c87816a321e64bacdf">[ed361bc515]</a> Sungtae Kim -- realtime: Increased reg_server character size</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29055">ASTERISK-29055</a>: Create a Bridge with video_single mode<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=519bb01d5f94b34a5a088dd570725aee3c5c8da2">[519bb01d5f]</a> Sungtae Kim -- res_stasis.c: Added video_single option for bridge creation</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26424">ASTERISK-26424</a>: app_voicemail: Undocumented behavior from VMSayName<br/>Reported by: Eric Smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb5ca4630f3a8a669286d668842ae3759488060a">[fb5ca4630f]</a> Sean Bright -- app_voicemail.c: Document VMSayName interruption behavior</li>
</ul><br><h4>Category: Configs/Samples</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29123">ASTERISK-29123</a>: logger.conf.sample missing comment mark on line 115<br/>Reported by: Andrew Siplas<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa5577fecdd470237ff395649b1b759d1f4e1712">[aa5577fecd]</a> Andrew Siplas -- logger.conf.sample: add missing comment mark</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29091">ASTERISK-29091</a>: Crash when ast_translator_build_path fails<br/>Reported by: Jasper van der Neut<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=47deb6967aa3e2a300460036f6f1ed54c1e807aa">[47deb6967a]</a> Jasper van der Neut -- channels: Don't dereference NULL pointer</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28430">ASTERISK-28430</a>: res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF<br/>Reported by: under<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e701ff95749fc74a7fabad6c4f409f470a5da1a">[5e701ff957]</a> Sean Bright -- tcptls.c: Don't close TCP client file descriptors more than once</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28311">ASTERISK-28311</a>: dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format<br/>Reported by: 周家建<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b370b70a704fb26b8b10417246c88d17f5c5b976">[b370b70a70]</a> Sean Bright -- dsp.c: Update calls to ast_format_cmp to check result properly</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28416">ASTERISK-28416</a>: Unable to get rtp codec payload code for slin<br/>Reported by: Brian J. Murrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b74e65f1116eec7b962eaa4f1608f77bb9e436cf">[b74e65f111]</a> Sean Bright -- format_cap: Perform codec lookups by pointer instead of name</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26424">ASTERISK-26424</a>: app_voicemail: Undocumented behavior from VMSayName<br/>Reported by: Eric Smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb5ca4630f3a8a669286d668842ae3759488060a">[fb5ca4630f]</a> Sean Bright -- app_voicemail.c: Document VMSayName interruption behavior</li>
</ul><br><h4>Category: Functions/func_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29085">ASTERISK-29085</a>: func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT<br/>Reported by: Péter Juhász<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7bd079e542c5ddc5727f5110355d32709ceb2c55">[7bd079e542]</a> Sean Bright -- func_curl.c: Prevent crash when using CURLOPT(httpheader)</li>
</ul><br><h4>Category: Resources/res_ari_endpoints</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29108">ASTERISK-29108</a>: resource_endpoints.c : Memory leak if endpoint not found<br/>Reported by: Jean Aunis - Prescom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b11f1242c5b3f36a27e35e40771d3f4ed5ab24d3">[b11f1242c5]</a> Jean Aunis -- resource_endpoints.c: memory leak when providing a 404 response</li>
</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29099">ASTERISK-29099</a>: res_musiconhold: Realtime MOH only loads a single entry<br/>Reported by: laszlovl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9511e8d379b5a8183eb77a4f2b0b982e919863f0">[9511e8d379]</a> laszlovl -- res_musiconhold: Load all realtime entries, not just the first</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24329">ASTERISK-24329</a>: Music On Hold announcement cuts intro of music the first time it is played<br/>Reported by: Thomas Frederiksen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ba11fed7e47c660c34e4fcdfbf05d42130516da">[6ba11fed7e]</a> Sean Bright -- res_musiconhold: Start playlist after initial announcement</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29013">ASTERISK-29013</a>: res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies<br/>Reported by: Sebastian Damm<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32c5ffc354495e24fd7165ee7daf9f66fd12bb93">[32c5ffc354]</a> Ben Ford -- AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29124">ASTERISK-29124</a>: res_pjsip: flow transport broken for outbound requests<br/>Reported by: Nick French<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7835bcf13ebf9340dc41abd16ab36334f6534ac">[f7835bcf13]</a> Nick French -- res_pjsip_session: Restore calls to ast_sip_message_apply_transport()</li>
</ul><br><h4>Category: Resources/res_pjsip_authenticator_digest</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29013">ASTERISK-29013</a>: res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies<br/>Reported by: Sebastian Damm<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32c5ffc354495e24fd7165ee7daf9f66fd12bb93">[32c5ffc354]</a> Ben Ford -- AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.</li>
</ul><br><h4>Category: Resources/res_pjsip_config_wizard</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29097">ASTERISK-29097</a>: res_pjsip_config_wizard: Crash when freeing string when failing to add extension<br/>Reported by: Vieri<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b11f143d5348a7a2b478aebbcb73402c40e34f2">[1b11f143d5]</a> Sean Bright -- pbx.c: On error, ast_add_extension2_lockopt should always free 'data'</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29051">ASTERISK-29051</a>: res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used<br/>Reported by: Sebastian Damm<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c1d9d4730bc5cac25f1fd653f7df6a49c5b2a9d">[0c1d9d4730]</a> Holger Hans Peter Freyther -- res_pjsip_sdp_rtp: Fix accidentally native bridging calls</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29014">ASTERISK-29014</a>: res_pjsip_session: Re-INVITE collisions aren't handled correctly<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a92119306ff7da899da4d528fa1cfd8efd67c998">[a92119306f]</a> George Joseph -- res_pjsip_session: Fix issue with COLP and 491</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3c67d585f016fc56d4e171ee217dc188655c44b">[d3c67d585f]</a> George Joseph -- res_pjsip_session: Handle multi-stream re-invites better</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29089">ASTERISK-29089</a>: RTP Ports not cleared after hangup<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bf404e15d255d07e767f86f36575577b8520f31">[1bf404e15d]</a> Joshua C. Colp -- res_pjsip_session: Fix session reference leak.</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29081">ASTERISK-29081</a>: res_stasis: Add compare function for bridges moh container<br/>Reported by: Hajek Michal<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4128bdfdb90c20bead90f22f356881209a094470">[4128bdfdb9]</a> Michal Hajek -- res_stasis.c: Add compare function for bridges moh container</li>
</ul><br><h3>New Feature</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29027">ASTERISK-29027</a>: Implement support for History-Info<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71820ea1141e93d64b5d5e2dc38472bc251710c1">[71820ea114]</a> Torrey Searle -- res_pjsip_diversion: implement support for History-Info</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=baf66a5816acd19f66d84760fb6bf8ad8e17eada">baf66a5816</a></td><td>Asterisk Development Team</td><td>Update for 17.9.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0654e5f77ba7859813f2495eda78d78e6032a3f3">0654e5f77b</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.9.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=132e2fd6ad77a76c454949b642311f88e282fcc7">132e2fd6ad</a></td><td>Alexander Traud</td><td>chan_sip: On authentication, pick MD5 for sure.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f198c85157da3302ab7b0b279fcb78996dfba929">f198c85157</a></td><td>Walter Doekes</td><td>main/say: Work around gcc 9 format-truncation false positive</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f5375261922f51b4cbbbcd391e7f8bed2fc5463">7f53752619</a></td><td>Kevin Harwell</td><td>res_pjsip, res_pjsip_session: initialize local variables</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=91dd18afc180db0bf5dc67befb7772ae2fe344de">91dd18afc1</a></td><td>Alexander Traud</td><td>install_prereq: Add GMime 3.0.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c363b4e964e8fecf2a470cfc6500ea70dba5f9e">2c363b4e96</a></td><td>Alexander Traud</td><td>BuildSystem: Enable Lua 5.4.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3efd0bab98aee692a5a13bb62293010cb2ce860">a3efd0bab9</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=680e07d514937b51304268ac65a0a4cb13a9684a">680e07d514</a></td><td>Joshua C. Colp</td><td>asterisk: Add verbose message stating support status.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0bc136c3ab42a302f98027552e2348afd4f4d61c">0bc136c3ab</a></td><td>George Joseph</td><td>app_confbridge/bridge_softmix: Add ability to force estimated bitrate</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1746703d2561138255739f57b04f2f9b3bed95c3">1746703d25</a></td><td>Torrey Searle</td><td>res_pjsip_diversion: fix double 181</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a68e33d91b621920086e0ace567f4408d98f356e">a68e33d91b</a></td><td>Sean Bright</td><td>res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26873338d7b364d521ab3c82a0ecc8d6766bd932">26873338d7</a></td><td>Joshua C. Colp</td><td>res_pjsip_session: Fix stream name memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78d9f6f74e2999a2fd9bb205af00bc424e2cc3c6">78d9f6f74e</a></td><td>George Joseph</td><td>logger.h: Fix ast_trace to respect scope_level</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29384f17c4aa90eeafcb8aae36fa878f93ce4c6a">29384f17c4</a></td><td>George Joseph</td><td>bridge_softmix/sfu_topologies_on_join: Ignore topology change failures</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=617f62b6ab3aa90118cf86f9f67a3db7725ad378">617f62b6ab</a></td><td>Sean Bright</td><td>res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1cd184250f75a6f9df70f2d10d1311a2860a6c06">1cd184250f</a></td><td>George Joseph</td><td>debugging: Add enough to choke a mule</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2469c94a7afdfc283aa765c3205254e1c716109c">2469c94a7a</a></td><td>Ben Ford</td><td>Bridging: Use a ref to bridge_channel's channel to prevent crash.</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-17.8.0-summary.html | 74
asterisk-17.8.0-summary.txt | 243 -
b/.version | 2
b/CHANGES | 12
b/ChangeLog | 575 ++
b/apps/app_confbridge.c | 3
b/apps/app_voicemail.c | 3
b/apps/confbridge/conf_config_parser.c | 21
b/apps/confbridge/include/confbridge.h | 2
b/asterisk-17.9.0-rc1-summary.html | 138
b/asterisk-17.9.0-rc1-summary.txt | 390 ++
b/bridges/bridge_softmix.c | 153
b/channels/chan_pjsip.c | 74
b/channels/chan_sip.c | 9
b/configs/samples/confbridge.conf.sample | 7
b/configs/samples/musiconhold.conf.sample | 4
b/configure | 2
b/configure.ac | 2
b/contrib/ast-db-manage/config/versions/1ae0609b6646_increse_reg_server_size.py | 22
b/contrib/ast-db-manage/config/versions/e658c26033ca_create_history_info_flag.py | 38
b/contrib/realtime/mysql/mysql_config.sql | 12
b/contrib/realtime/postgresql/postgresql_config.sql | 12
b/contrib/scripts/install_prereq | 2
b/funcs/func_curl.c | 7
b/include/asterisk/bridge.h | 14
b/include/asterisk/bridge_channel.h | 14
b/include/asterisk/format_cache.h | 13
b/include/asterisk/logger.h | 4
b/include/asterisk/pbx.h | 8
b/include/asterisk/res_pjsip.h | 48
b/include/asterisk/res_pjsip_session.h | 8
b/include/asterisk/stream.h | 4
b/main/asterisk.c | 10
b/main/bridge.c | 44
b/main/bridge_channel.c | 20
b/main/channel.c | 14
b/main/dsp.c | 4
b/main/format_cache.c | 21
b/main/format_cap.c | 2
b/main/pbx.c | 12
b/main/say.c | 20
b/main/stream.c | 30
b/main/tcptls.c | 12
b/res/ari/resource_bridges.h | 4
b/res/ari/resource_endpoints.c | 1
b/res/parking/parking_bridge_features.c | 1
b/res/res_musiconhold.c | 24
b/res/res_parking.c | 1
b/res/res_pjsip.c | 54
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_transport_management.c | 2
b/res/res_pjsip_config_wizard.c | 1
b/res/res_pjsip_diversion.c | 326 +
b/res/res_pjsip_pubsub.c | 10
b/res/res_pjsip_sdp_rtp.c | 3
b/res/res_pjsip_session.c | 1939 ++++++++--
b/res/res_stasis.c | 31
b/res/stasis/stasis_bridge.c | 2
b/rest-api/api-docs/bridges.json | 4
59 files changed, 3729 insertions(+), 784 deletions(-)</pre><br></html>

391
asterisk-17.9.0-summary.txt Normal file
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@@ -0,0 +1,391 @@
Release Summary
asterisk-17.9.0
Date: 2020-11-19
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-17.8.0.
----------------------------------------------------------------------
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
9 Sean Bright 2 Sebastian Damm
6 George Joseph 2 George Joseph
3 Asterisk Development Team 2 sungtae kim
3 Joshua C. Colp 1 Sandro Gauci
3 Alexander Traud 1 Nick French
2 Kevin Harwell 1 Michal Hajek
2 Torrey Searle 1 Ross Beer
2 Ben Ford 1 under
2 Sungtae Kim 1 å*¨å®¶å»º
1 Holger Hans Peter Freyther 1 Torrey Searle
1 Walter Doekes 1 Hajek Michal
1 Michal Hajek 1 Vieri
1 Jean Aunis 1 Péter Juhász
1 Nick French 1 Andrew Siplas
1 laszlovl 1 Jean Aunis - Prescom
1 Jasper van der Neut 1 laszlovl
1 Andrew Siplas 1 Thomas Frederiksen
1 Eric Smith
1 Brian J. Murrell
1 Jasper van der Neut
----------------------------------------------------------------------
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: pjproject/pjsip
ASTERISK-29057: pjsip: Crash on call rejection during high load
Reported by: Sandro Gauci
* [fab1ef83b2] Kevin Harwell -- AST-2020-001 - res_pjsip: Return dialog
locked and referenced
Improvement
Category: Resources/General
ASTERISK-29056: Increase reg_server column size for ps_contacts table
realtime
Reported by: sungtae kim
* [ed361bc515] Sungtae Kim -- realtime: Increased reg_server character
size
Category: Resources/res_stasis
ASTERISK-29055: Create a Bridge with video_single mode
Reported by: sungtae kim
* [519bb01d5f] Sungtae Kim -- res_stasis.c: Added video_single option
for bridge creation
Bug
Category: Applications/app_voicemail
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [fb5ca4630f] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
Category: Configs/Samples
ASTERISK-29123: logger.conf.sample missing comment mark on line 115
Reported by: Andrew Siplas
* [aa5577fecd] Andrew Siplas -- logger.conf.sample: add missing comment
mark
Category: Core/Channels
ASTERISK-29091: Crash when ast_translator_build_path fails
Reported by: Jasper van der Neut
* [47deb6967a] Jasper van der Neut -- channels: Don't dereference NULL
pointer
Category: Core/General
ASTERISK-28430: res_rtp_asterisk.c: FRACK!, Failed assertion errno !=
EBADF
Reported by: under
* [5e701ff957] Sean Bright -- tcptls.c: Don't close TCP client file
descriptors more than once
ASTERISK-28311: dsp: ast_dsp_silence_noise_with_energy wrong judgment of
frame format
Reported by: å*¨å®¶å»º
* [b370b70a70] Sean Bright -- dsp.c: Update calls to ast_format_cmp to
check result properly
Category: Core/RTP
ASTERISK-28416: Unable to get rtp codec payload code for slin
Reported by: Brian J. Murrell
* [b74e65f111] Sean Bright -- format_cap: Perform codec lookups by
pointer instead of name
Category: Documentation
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [fb5ca4630f] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
Category: Functions/func_curl
ASTERISK-29085: func_curl: Segmentation fault when using CURL after
setting httpheader CURLOPT
Reported by: Péter Juhász
* [7bd079e542] Sean Bright -- func_curl.c: Prevent crash when using
CURLOPT(httpheader)
Category: Resources/res_ari_endpoints
ASTERISK-29108: resource_endpoints.c : Memory leak if endpoint not found
Reported by: Jean Aunis - Prescom
* [b11f1242c5] Jean Aunis -- resource_endpoints.c: memory leak when
providing a 404 response
Category: Resources/res_musiconhold
ASTERISK-29099: res_musiconhold: Realtime MOH only loads a single entry
Reported by: laszlovl
* [9511e8d379] laszlovl -- res_musiconhold: Load all realtime entries,
not just the first
ASTERISK-24329: Music On Hold announcement cuts intro of music the first
time it is played
Reported by: Thomas Frederiksen
* [6ba11fed7e] Sean Bright -- res_musiconhold: Start playlist after
initial announcement
Category: Resources/res_pjsip
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with
auth) on 407 replies
Reported by: Sebastian Damm
* [32c5ffc354] Ben Ford -- AST-2020-002 - res_pjsip: Stop sending
INVITEs after challenge limit.
ASTERISK-29124: res_pjsip: flow transport broken for outbound requests
Reported by: Nick French
* [f7835bcf13] Nick French -- res_pjsip_session: Restore calls to
ast_sip_message_apply_transport()
Category: Resources/res_pjsip_authenticator_digest
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with
auth) on 407 replies
Reported by: Sebastian Damm
* [32c5ffc354] Ben Ford -- AST-2020-002 - res_pjsip: Stop sending
INVITEs after challenge limit.
Category: Resources/res_pjsip_config_wizard
ASTERISK-29097: res_pjsip_config_wizard: Crash when freeing string when
failing to add extension
Reported by: Vieri
* [1b11f143d5] Sean Bright -- pbx.c: On error,
ast_add_extension2_lockopt should always free 'data'
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-29051: res_pjsip_sdp_rtp: Does not set correct values on RTP
instance when "auto" DTMF is used
Reported by: Sebastian Damm
* [0c1d9d4730] Holger Hans Peter Freyther -- res_pjsip_sdp_rtp: Fix
accidentally native bridging calls
Category: Resources/res_pjsip_session
ASTERISK-29014: res_pjsip_session: Re-INVITE collisions aren't handled
correctly
Reported by: George Joseph
* [a92119306f] George Joseph -- res_pjsip_session: Fix issue with COLP
and 491
* [d3c67d585f] George Joseph -- res_pjsip_session: Handle multi-stream
re-invites better
Category: Resources/res_rtp_asterisk
ASTERISK-29089: RTP Ports not cleared after hangup
Reported by: Ross Beer
* [1bf404e15d] Joshua C. Colp -- res_pjsip_session: Fix session
reference leak.
Category: Resources/res_stasis
ASTERISK-29081: res_stasis: Add compare function for bridges moh container
Reported by: Hajek Michal
* [4128bdfdb9] Michal Hajek -- res_stasis.c: Add compare function for
bridges moh container
New Feature
Category: Resources/res_pjsip_diversion
ASTERISK-29027: Implement support for History-Info
Reported by: Torrey Searle
* [71820ea114] Torrey Searle -- res_pjsip_diversion: implement support
for History-Info
----------------------------------------------------------------------
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 |
|------------+---------------+-------------------------------------------|
| | Asterisk | |
| baf66a5816 | Development | Update for 17.9.0-rc1 |
| | Team | |
|------------+---------------+-------------------------------------------|
| | Asterisk | |
| 0654e5f77b | Development | Update CHANGES and UPGRADE.txt for 17.9.0 |
| | Team | |
|------------+---------------+-------------------------------------------|
| 132e2fd6ad | Alexander | chan_sip: On authentication, pick MD5 for |
| | Traud | sure. |
|------------+---------------+-------------------------------------------|
| f198c85157 | Walter Doekes | main/say: Work around gcc 9 |
| | | format-truncation false positive |
|------------+---------------+-------------------------------------------|
| 7f53752619 | Kevin Harwell | res_pjsip, res_pjsip_session: initialize |
| | | local variables |
|------------+---------------+-------------------------------------------|
| 91dd18afc1 | Alexander | install_prereq: Add GMime 3.0. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| 2c363b4e96 | Alexander | BuildSystem: Enable Lua 5.4. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| | Asterisk | |
| a3efd0bab9 | Development | Update CHANGES and UPGRADE.txt for 17.8.0 |
| | Team | |
|------------+---------------+-------------------------------------------|
| 680e07d514 | Joshua C. | asterisk: Add verbose message stating |
| | Colp | support status. |
|------------+---------------+-------------------------------------------|
| 0bc136c3ab | George Joseph | app_confbridge/bridge_softmix: Add |
| | | ability to force estimated bitrate |
|------------+---------------+-------------------------------------------|
| 1746703d25 | Torrey Searle | res_pjsip_diversion: fix double 181 |
|------------+---------------+-------------------------------------------|
| a68e33d91b | Sean Bright | res_musiconhold: Clarify that playlist |
| | | mode only supports HTTP(S) URLs |
|------------+---------------+-------------------------------------------|
| 26873338d7 | Joshua C. | res_pjsip_session: Fix stream name memory |
| | Colp | leak. |
|------------+---------------+-------------------------------------------|
| 78d9f6f74e | George Joseph | logger.h: Fix ast_trace to respect |
| | | scope_level |
|------------+---------------+-------------------------------------------|
| 29384f17c4 | George Joseph | bridge_softmix/sfu_topologies_on_join: |
| | | Ignore topology change failures |
|------------+---------------+-------------------------------------------|
| 617f62b6ab | Sean Bright | res_pjsip_session.c: Fix build when |
| | | TEST_FRAMEWORK is not defined |
|------------+---------------+-------------------------------------------|
| 1cd184250f | George Joseph | debugging: Add enough to choke a mule |
|------------+---------------+-------------------------------------------|
| 2469c94a7a | Ben Ford | Bridging: Use a ref to bridge_channel's |
| | | channel to prevent crash. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-17.8.0-summary.html | 74
asterisk-17.8.0-summary.txt | 243 -
b/.version | 2
b/CHANGES | 12
b/ChangeLog | 575 ++
b/apps/app_confbridge.c | 3
b/apps/app_voicemail.c | 3
b/apps/confbridge/conf_config_parser.c | 21
b/apps/confbridge/include/confbridge.h | 2
b/asterisk-17.9.0-rc1-summary.html | 138
b/asterisk-17.9.0-rc1-summary.txt | 390 ++
b/bridges/bridge_softmix.c | 153
b/channels/chan_pjsip.c | 74
b/channels/chan_sip.c | 9
b/configs/samples/confbridge.conf.sample | 7
b/configs/samples/musiconhold.conf.sample | 4
b/configure | 2
b/configure.ac | 2
b/contrib/ast-db-manage/config/versions/1ae0609b6646_increse_reg_server_size.py | 22
b/contrib/ast-db-manage/config/versions/e658c26033ca_create_history_info_flag.py | 38
b/contrib/realtime/mysql/mysql_config.sql | 12
b/contrib/realtime/postgresql/postgresql_config.sql | 12
b/contrib/scripts/install_prereq | 2
b/funcs/func_curl.c | 7
b/include/asterisk/bridge.h | 14
b/include/asterisk/bridge_channel.h | 14
b/include/asterisk/format_cache.h | 13
b/include/asterisk/logger.h | 4
b/include/asterisk/pbx.h | 8
b/include/asterisk/res_pjsip.h | 48
b/include/asterisk/res_pjsip_session.h | 8
b/include/asterisk/stream.h | 4
b/main/asterisk.c | 10
b/main/bridge.c | 44
b/main/bridge_channel.c | 20
b/main/channel.c | 14
b/main/dsp.c | 4
b/main/format_cache.c | 21
b/main/format_cap.c | 2
b/main/pbx.c | 12
b/main/say.c | 20
b/main/stream.c | 30
b/main/tcptls.c | 12
b/res/ari/resource_bridges.h | 4
b/res/ari/resource_endpoints.c | 1
b/res/parking/parking_bridge_features.c | 1
b/res/res_musiconhold.c | 24
b/res/res_parking.c | 1
b/res/res_pjsip.c | 54
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_transport_management.c | 2
b/res/res_pjsip_config_wizard.c | 1
b/res/res_pjsip_diversion.c | 326 +
b/res/res_pjsip_pubsub.c | 10
b/res/res_pjsip_sdp_rtp.c | 3
b/res/res_pjsip_session.c | 1939 ++++++++--
b/res/res_stasis.c | 31
b/res/stasis/stasis_bridge.c | 2
b/rest-api/api-docs/bridges.json | 4
59 files changed, 3729 insertions(+), 784 deletions(-)

View File

@@ -0,0 +1,41 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start DATETIME,
answer DATETIME,
end DATETIME,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';

File diff suppressed because it is too large Load Diff

View File

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

View File

@@ -0,0 +1,45 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL,
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start TIMESTAMP WITHOUT TIME ZONE,
answer TIMESTAMP WITHOUT TIME ZONE,
"end" TIMESTAMP WITHOUT TIME ZONE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
-- Running upgrade 210693f3123d -> 54cde9847798
ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
COMMIT;

File diff suppressed because it is too large Load Diff

View File

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