Update for 18.1.1

18.1
Asterisk Development Team 4 years ago
parent e5a48b86c9
commit c48ddab47b

@ -1 +1 @@
18.1.0
18.1.1

@ -1,3 +1,19 @@
2020-12-22 21:10 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 18.1.1 Released.
2020-12-22 02:58 +0000 [e5a48b86c9] Torrey Searle <tsearle@voxbone.com>
* res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
Add a check to see if the URI is a Tel URI and prevent crashing on
trying to retrieve the reason parameter.
ASTERISK-29191
ASTERISK-29219
Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
2020-11-19 12:39 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 18.1.0 Released.

@ -1,188 +0,0 @@
<!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-18.1.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.1.0</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-18.0.0.</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%">12 Sean Bright <sean.bright@gmail.com><br/>8 George Joseph <gjoseph@digium.com><br/>8 Alexander Traud <pabstraud@compuserve.com><br/>3 Kevin Harwell <kharwell@sangoma.com><br/>3 Joshua C. Colp <jcolp@sangoma.com><br/>2 Torrey Searle <tsearle@voxbone.com><br/>2 Asterisk Development Team <asteriskteam@digium.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 Dovid Bender <dovid@telecurve.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Alexei Gradinari <alex2grad@gmail.com><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%">3 Alexander Traud <pabstraud@compuserve.com><br/>2 George Joseph <gjoseph@digium.com><br/>2 sungtae kim <pchero21@gmail.com><br/>2 Ross Beer <ross.beer@voicehost.co.uk><br/>2 Sebastian Damm <damm@sipgate.de><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 周家建 <zhou_0611@163.com><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Kevin Harwell <kharwell@digium.com><br/>1 dovid <dovi5988@dovid.net><br/>1 Brian J. Murrell <brian@interlinx.bc.ca><br/>1 Vieri <vieridipaola@gmail.com><br/>1 Nick French <nickfrench@gmail.com><br/>1 Walter Doekes<br/>1 Benjamin M. <mailinglist@perspectives.qc.ca><br/>1 Péter Juhász <peter.juhasz@comnica.com><br/>1 under <under@list.ru><br/>1 Thomas Frederiksen <tommer@nicesurprise.com><br/>1 Michal Hajek <michal.hajek@daktela.com><br/>1 Michael Newton <miken32@gmail.com><br/>1 Sandro Gauci <sandro@enablesecurity.com><br/>1 Dovid Bender<br/>1 Jasper van der Neut <jasper@isotopic.nl><br/>1 Torrey Searle <tsearle@gmail.com><br/>1 laszlovl <digium@lvlconsultancy.nl><br/>1 Hajek Michal <michal.hajek@daktela.com><br/>1 Eric Smith <abkowald@gmail.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>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=6baa4b53bef5d9c53692f22cf146215b42de1e89">[6baa4b53be]</a> Kevin Harwell -- AST-2020-001 - res_pjsip: Return dialog locked and referenced</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=83140c9fed442719a150dd89c1717808d00e9709">[83140c9fed]</a> Torrey Searle -- res_pjsip_diversion: implement support for History-Info</li>
</ul><br><h3>Bug</h3><h4>Category: . I did not set the category correctly.</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29146">ASTERISK-29146</a>: GCC Warnings: %s directive argument is null.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f86af1fbd05be587f2f9a211977b63a9c7527458">[f86af1fbd0]</a> Alexander Traud -- Compiler fixes for GCC when printf %s is NULL</li>
</ul><br><h4>Category: Applications/app_directory</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29144">ASTERISK-29144</a>: GCC Warnings with OPTIMIZE=-Og make<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0ee53dc9cef33619f5f0fd5efffb71dba6455ed">[e0ee53dc9c]</a> Alexander Traud -- Compiler fixes for GCC with -Og</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29144">ASTERISK-29144</a>: GCC Warnings with OPTIMIZE=-Og make<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0ee53dc9cef33619f5f0fd5efffb71dba6455ed">[e0ee53dc9c]</a> Alexander Traud -- Compiler fixes for GCC with -Og</li>
</ul><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=abee490639d31cd8d571b7163a1a542df2a1dcfb">[abee490639]</a> Sean Bright -- app_voicemail.c: Document VMSayName interruption behavior</li>
</ul><br><h4>Category: Channels/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29144">ASTERISK-29144</a>: GCC Warnings with OPTIMIZE=-Og make<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0ee53dc9cef33619f5f0fd5efffb71dba6455ed">[e0ee53dc9c]</a> Alexander Traud -- Compiler fixes for GCC with -Og</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=ff33f7f44fc5918369f1c1d20521fd1769f86dea">[ff33f7f44f]</a> Andrew Siplas -- logger.conf.sample: add missing comment mark</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29142">ASTERISK-29142</a>: sip_to_pjsip.py: doesn't read globbed includes<br/>Reported by: Michael Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe540d03261aa07d2f341f688966d41f24702a90">[fe540d0326]</a> Sean Bright -- sip_to_pjsip.py: Handle #include globs and other fixes</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=08ccfd4588765eaae76ab3c3734a4cfd74138160">[08ccfd4588]</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=a6faa53af0540483b56a42eb5ef67d2d4b5753be">[a6faa53af0]</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=9b08eddf90d05b9a701f1326b9d26f23f72e38de">[9b08eddf90]</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=49643029849b366074df0291fd5f8a859c970c8a">[4964302984]</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-29136">ASTERISK-29136</a>: config: Sample features.conf incorrectly includes " around sound files<br/>Reported by: Benjamin M.<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6f321b561ae1c87e495c4e711ce2a948f7db5dca">[6f321b561a]</a> Sean Bright -- features.conf.sample: Sample sound files incorrectly quoted</li>
</ul><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=abee490639d31cd8d571b7163a1a542df2a1dcfb">[abee490639]</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-28825">ASTERISK-28825</a>: Any curl response checks out as valid even if 404 is returned.<br/>Reported by: dovid<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c635c782654f357999a7330f17b421303a878636">[c635c78265]</a> Dovid Bender -- func_curl.c: Allow user to set what return codes constitute a failure.</li>
</ul><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=28c88e8fe2c0edf26821a998681deed6741564c7">[28c88e8fe2]</a> Sean Bright -- func_curl.c: Prevent crash when using CURLOPT(httpheader)</li>
</ul><br><h4>Category: Functions/func_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29144">ASTERISK-29144</a>: GCC Warnings with OPTIMIZE=-Og make<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0ee53dc9cef33619f5f0fd5efffb71dba6455ed">[e0ee53dc9c]</a> Alexander Traud -- Compiler fixes for GCC with -Og</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=7ced14486795342d8501f8ba10306841286715d2">[7ced144867]</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=b3b6b5e9f77852066a7c902acd77a586214e0773">[b3b6b5e9f7]</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=d0644faa5abbe10c892aade339fc694d0c492c34">[d0644faa5a]</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-28933">ASTERISK-28933</a>: res_pjsip.so fails to load when bundled pjproject is compiled without libssl<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a6037778b4f661e1eab78e07ce46f94a1ed9533">[5a6037778b]</a> Alexander Traud -- res_pjsip/config_transport: Load and run without OpenSSL.</li>
</ul><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=82325ba58bd29efea4f84e37b014747049cf6dff">[82325ba58b]</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=f041763e3b832c5b9db83c60e1956b91d846666d">[f041763e3b]</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=82325ba58bd29efea4f84e37b014747049cf6dff">[82325ba58b]</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=5a0b19a4f37a22a5cb6ccb76c631a91eb5e823fe">[5a0b19a4f3]</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=4499fbc81964a22e93676eb55801e9abe4e3ccd0">[4499fbc819]</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-29109">ASTERISK-29109</a>: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=412b385de5cb1c124daa3e79e19b7ae0af190d63">[412b385de5]</a> Joshua C. Colp -- res_pjsip: Adjust outgoing offer call pref.</li>
</ul><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=cc71be007870bb19856aa9559cd6485d45c4edb5">[cc71be0078]</a> George Joseph -- res_pjsip_session: Fix issue with COLP and 491</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4f3b17dd3ee3836270af2b6870afebe7b7c0e3c">[d4f3b17dd3]</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=957aff751d1363953a00aac7ddd9c772b405c574">[957aff751d]</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=2bce21da88b63105b87a48443a2a353b690ed332">[2bce21da88]</a> Michal Hajek -- res_stasis.c: Add compare function for bridges moh container</li>
</ul><br><h4>Category: Utilities/muted</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29145">ASTERISK-29145</a>: GCC Warnings with OPTIMIZE=-Os make<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2dacadd9dfca2b1ef1fca52f4dbd69968cbf2783">[2dacadd9df]</a> Alexander Traud -- Compiler fixes for GCC with -Os</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29054">ASTERISK-29054</a>: Logger: Add debug logging categories<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6255e7976c23d86c34a28f42557bd9030282be3a">[6255e7976c]</a> Kevin Harwell -- Logging: Add debug logging categories</li>
</ul><br><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=1fd12b88c7a7a833f084db778af1ee8d5f38601d">[1fd12b88c7]</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=a0d41a27d49c2de03eab4572d0bcc721a8f93dd4">[a0d41a27d4]</a> Sungtae Kim -- res_stasis.c: Added video_single option for bridge creation</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=50145c837a330e16a9e9b91bd1f3639d304a5f81">50145c837a</a></td><td>Asterisk Development Team</td><td>Update for 18.1.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98d1537c1eab8334416238091632025325297319">98d1537c1e</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.1.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=860e40dd80f0603582b98a7da8150f15b564cce3">860e40dd80</a></td><td>George Joseph</td><td>res_pjsip_outbound_registration.c: Use our own scheduler and other stuff</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=569fc289664f70190c8ee6884b44a15225987c24">569fc28966</a></td><td>George Joseph</td><td>pjsip_scheduler.c: Add type ONESHOT and enhance cli show command</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da0f2ea99e178b958e1371484e6b115e3e3e3da7">da0f2ea99e</a></td><td>Alexei Gradinari</td><td>sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be54c7e9ea29d90267e671c6d7c718a44ec1c6ea">be54c7e9ea</a></td><td>Alexander Traud</td><td>res_stir_shaken: Include OpenSSL headers where used actually.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b25c75d7be8c9a09f791a0fa79900396d00465a">5b25c75d7b</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=fb3b14ab7db9026bca39f82931564350b401c557">fb3b14ab7d</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=439f7bb8485c08e2825e954158b8f8e898099ff9">439f7bb848</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=f89531cb98f59a5bf84067026e8676cdbb306079">f89531cb98</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=2773f931548c18bfcf33197a66176988a8903330">2773f93154</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=4a049ad51055430dc84f2217a7ca240f92f791e7">4a049ad510</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=c470327e6ca239ce4ddc74600a34484276e928b0">c470327e6c</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=5929e0ccbd1cbcca41b47cd21de1e0c5406fbfbc">5929e0ccbd</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=9eeb40af33437f50240b9eb61d9ab8756fa7a0aa">9eeb40af33</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=99bd7d95de54cf23446050435024295743be329c">99bd7d95de</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=c90c182932a123f470e3df100a89ff163aca5526">c90c182932</a></td><td>Sean Bright</td><td>audiosocket: Fix module menuselect descriptions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fdc13060df8f79cd5277fb2c8a6c70d22b5cb69c">fdc13060df</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=6f32c254bea47ed54e810dfee5305b022453cac9">6f32c254be</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=ad4f2a8c99d3812868e0e094735df45c3970943d">ad4f2a8c99</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=7eaae4e7b600ec0aecd6e30b75d89c99241d31fa">7eaae4e7b6</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-18.0.0-summary.html | 1162 ----
asterisk-18.0.0-summary.txt | 2873 ----------
b/.version | 2
b/CHANGES | 30
b/ChangeLog | 742 ++
b/addons/ooh323c/src/ooq931.c | 2
b/apps/app_confbridge.c | 3
b/apps/app_directory.c | 2
b/apps/app_voicemail.c | 5
b/apps/confbridge/conf_config_parser.c | 21
b/apps/confbridge/include/confbridge.h | 2
b/asterisk-18.1.0-rc1-summary.html | 188
b/asterisk-18.1.0-rc1-summary.txt | 496 +
b/bridges/bridge_softmix.c | 153
b/channels/chan_audiosocket.c | 5
b/channels/chan_iax2.c | 2
b/channels/chan_pjsip.c | 94
b/channels/chan_sip.c | 13
b/configs/samples/confbridge.conf.sample | 7
b/configs/samples/features.conf.sample | 4
b/configs/samples/musiconhold.conf.sample | 4
b/configs/samples/res_curl.conf.sample | 1
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/contrib/scripts/sip_to_pjsip/astconfigparser.py | 43
b/funcs/func_curl.c | 48
b/funcs/func_odbc.c | 2
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/logger_category.h | 178
b/include/asterisk/pbx.h | 8
b/include/asterisk/res_pjsip.h | 94
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/res_stir_shaken.h | 3
b/include/asterisk/rtp_engine.h | 79
b/include/asterisk/sched.h | 5
b/include/asterisk/stream.h | 4
b/include/asterisk/stun.h | 25
b/main/bridge.c | 44
b/main/bridge_channel.c | 20
b/main/channel.c | 14
b/main/cli.c | 51
b/main/dsp.c | 4
b/main/format_cache.c | 21
b/main/format_cap.c | 2
b/main/indications.c | 6
b/main/logger.c | 5
b/main/logger_category.c | 324 +
b/main/pbx.c | 12
b/main/rtp_engine.c | 68
b/main/say.c | 20
b/main/stream.c | 30
b/main/stun.c | 61
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_audiosocket.c | 3
b/res/res_musiconhold.c | 24
b/res/res_parking.c | 1
b/res/res_pjsip.c | 56
b/res/res_pjsip/config_transport.c | 18
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_scheduler.c | 180
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_outbound_registration.c | 286
b/res/res_pjsip_pubsub.c | 10
b/res/res_pjsip_sdp_rtp.c | 26
b/res/res_pjsip_session.c | 1977 +++++-
b/res/res_pjsip_stir_shaken.c | 1
b/res/res_rtp_asterisk.c | 403 -
b/res/res_stasis.c | 31
b/res/res_stir_shaken.c | 4
b/res/res_stir_shaken/stir_shaken.c | 3
b/res/stasis/stasis_bridge.c | 2
84 files changed, 5520 insertions(+), 4976 deletions(-)</pre><br></html>

@ -1,499 +0,0 @@
Release Summary
asterisk-18.1.0
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-18.0.0.
----------------------------------------------------------------------
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
12 Sean Bright 3 Alexander Traud
8 George Joseph 2 George Joseph
8 Alexander Traud 2 sungtae kim
3 Kevin Harwell 2 Ross Beer
3 Joshua C. Colp 2 Sebastian Damm
2 Torrey Searle 1 Walter Doekes
2 Asterisk Development Team 1 å*¨å®¶å»º
2 Ben Ford 1 Andrew Siplas
2 Sungtae Kim 1 Jean Aunis - Prescom
1 Holger Hans Peter Freyther 1 Kevin Harwell
1 Dovid Bender 1 dovid
1 Walter Doekes 1 Brian J. Murrell
1 Alexei Gradinari 1 Vieri
1 Michal Hajek 1 Nick French
1 Jean Aunis 1 Walter Doekes
1 Nick French 1 Benjamin M.
1 laszlovl 1 Péter Juhász
1 Jasper van der Neut 1 under
1 Andrew Siplas 1 Thomas Frederiksen
1 Michal Hajek
1 Michael Newton
1 Sandro Gauci
1 Dovid Bender
1 Jasper van der Neut
1 Torrey Searle
1 laszlovl
1 Hajek Michal
1 Eric Smith
----------------------------------------------------------------------
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
* [6baa4b53be] Kevin Harwell -- AST-2020-001 - res_pjsip: Return dialog
locked and referenced
New Feature
Category: Resources/res_pjsip_diversion
ASTERISK-29027: Implement support for History-Info
Reported by: Torrey Searle
* [83140c9fed] Torrey Searle -- res_pjsip_diversion: implement support
for History-Info
Bug
Category: . I did not set the category correctly.
ASTERISK-29146: GCC Warnings: â**%sâ** directive argument is null.
Reported by: Alexander Traud
* [f86af1fbd0] Alexander Traud -- Compiler fixes for GCC when printf %s
is NULL
Category: Applications/app_directory
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [e0ee53dc9c] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Applications/app_voicemail
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [e0ee53dc9c] Alexander Traud -- Compiler fixes for GCC with -Og
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [abee490639] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
Category: Channels/General
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [e0ee53dc9c] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Configs/Samples
ASTERISK-29123: logger.conf.sample missing comment mark on line 115
Reported by: Andrew Siplas
* [ff33f7f44f] Andrew Siplas -- logger.conf.sample: add missing comment
mark
Category: Contrib/General
ASTERISK-29142: sip_to_pjsip.py: doesn't read globbed includes
Reported by: Michael Newton
* [fe540d0326] Sean Bright -- sip_to_pjsip.py: Handle #include globs and
other fixes
Category: Core/Channels
ASTERISK-29091: Crash when ast_translator_build_path fails
Reported by: Jasper van der Neut
* [08ccfd4588] 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
* [a6faa53af0] 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
* [9b08eddf90] 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
* [4964302984] Sean Bright -- format_cap: Perform codec lookups by
pointer instead of name
Category: Documentation
ASTERISK-29136: config: Sample features.conf incorrectly includes " around
sound files
Reported by: Benjamin M.
* [6f321b561a] Sean Bright -- features.conf.sample: Sample sound files
incorrectly quoted
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [abee490639] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
Category: Functions/func_curl
ASTERISK-28825: Any curl response checks out as valid even if 404 is
returned.
Reported by: dovid
* [c635c78265] Dovid Bender -- func_curl.c: Allow user to set what
return codes constitute a failure.
ASTERISK-29085: func_curl: Segmentation fault when using CURL after
setting httpheader CURLOPT
Reported by: Péter Juhász
* [28c88e8fe2] Sean Bright -- func_curl.c: Prevent crash when using
CURLOPT(httpheader)
Category: Functions/func_odbc
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [e0ee53dc9c] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Resources/res_ari_endpoints
ASTERISK-29108: resource_endpoints.c : Memory leak if endpoint not found
Reported by: Jean Aunis - Prescom
* [7ced144867] 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
* [b3b6b5e9f7] 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
* [d0644faa5a] Sean Bright -- res_musiconhold: Start playlist after
initial announcement
Category: Resources/res_pjsip
ASTERISK-28933: res_pjsip.so fails to load when bundled pjproject is
compiled without libssl
Reported by: Walter Doekes
* [5a6037778b] Alexander Traud -- res_pjsip/config_transport: Load and
run without OpenSSL.
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with
auth) on 407 replies
Reported by: Sebastian Damm
* [82325ba58b] 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
* [f041763e3b] 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
* [82325ba58b] 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
* [5a0b19a4f3] 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
* [4499fbc819] Holger Hans Peter Freyther -- res_pjsip_sdp_rtp: Fix
accidentally native bridging calls
Category: Resources/res_pjsip_session
ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due
to codec negotiation after upgrading from Asterisk 16
Reported by: Ross Beer
* [412b385de5] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call
pref.
ASTERISK-29014: res_pjsip_session: Re-INVITE collisions aren't handled
correctly
Reported by: George Joseph
* [cc71be0078] George Joseph -- res_pjsip_session: Fix issue with COLP
and 491
* [d4f3b17dd3] 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
* [957aff751d] 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
* [2bce21da88] Michal Hajek -- res_stasis.c: Add compare function for
bridges moh container
Category: Utilities/muted
ASTERISK-29145: GCC Warnings with OPTIMIZE=-Os make
Reported by: Alexander Traud
* [2dacadd9df] Alexander Traud -- Compiler fixes for GCC with -Os
Improvement
Category: Core/Logging
ASTERISK-29054: Logger: Add debug logging categories
Reported by: Kevin Harwell
* [6255e7976c] Kevin Harwell -- Logging: Add debug logging categories
Category: Resources/General
ASTERISK-29056: Increase reg_server column size for ps_contacts table
realtime
Reported by: sungtae kim
* [1fd12b88c7] 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
* [a0d41a27d4] Sungtae Kim -- res_stasis.c: Added video_single option
for bridge creation
----------------------------------------------------------------------
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 | |
| 50145c837a | Development | Update for 18.1.0-rc1 |
| | Team | |
|------------+---------------+-------------------------------------------|
| | Asterisk | |
| 98d1537c1e | Development | Update CHANGES and UPGRADE.txt for 18.1.0 |
| | Team | |
|------------+---------------+-------------------------------------------|
| 860e40dd80 | George Joseph | res_pjsip_outbound_registration.c: Use |
| | | our own scheduler and other stuff |
|------------+---------------+-------------------------------------------|
| 569fc28966 | George Joseph | pjsip_scheduler.c: Add type ONESHOT and |
| | | enhance cli show command |
|------------+---------------+-------------------------------------------|
| da0f2ea99e | Alexei | sched: AST_SCHED_REPLACE_UNREF can lead |
| | Gradinari | to use after free of data |
|------------+---------------+-------------------------------------------|
| be54c7e9ea | Alexander | res_stir_shaken: Include OpenSSL headers |
| | Traud | where used actually. |
|------------+---------------+-------------------------------------------|
| 5b25c75d7b | Alexander | chan_sip: On authentication, pick MD5 for |
| | Traud | sure. |
|------------+---------------+-------------------------------------------|
| fb3b14ab7d | Walter Doekes | main/say: Work around gcc 9 |
| | | format-truncation false positive |
|------------+---------------+-------------------------------------------|
| 439f7bb848 | Kevin Harwell | res_pjsip, res_pjsip_session: initialize |
| | | local variables |
|------------+---------------+-------------------------------------------|
| f89531cb98 | Alexander | install_prereq: Add GMime 3.0. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| 2773f93154 | Alexander | BuildSystem: Enable Lua 5.4. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| 4a049ad510 | George Joseph | app_confbridge/bridge_softmix: Add |
| | | ability to force estimated bitrate |
|------------+---------------+-------------------------------------------|
| c470327e6c | Torrey Searle | res_pjsip_diversion: fix double 181 |
|------------+---------------+-------------------------------------------|
| 5929e0ccbd | Sean Bright | res_musiconhold: Clarify that playlist |
| | | mode only supports HTTP(S) URLs |
|------------+---------------+-------------------------------------------|
| 9eeb40af33 | Joshua C. | res_pjsip_session: Fix stream name memory |
| | Colp | leak. |
|------------+---------------+-------------------------------------------|
| 99bd7d95de | George Joseph | logger.h: Fix ast_trace to respect |
| | | scope_level |
|------------+---------------+-------------------------------------------|
| c90c182932 | Sean Bright | audiosocket: Fix module menuselect |
| | | descriptions |
|------------+---------------+-------------------------------------------|
| fdc13060df | George Joseph | bridge_softmix/sfu_topologies_on_join: |
| | | Ignore topology change failures |
|------------+---------------+-------------------------------------------|
| 6f32c254be | Sean Bright | res_pjsip_session.c: Fix build when |
| | | TEST_FRAMEWORK is not defined |
|------------+---------------+-------------------------------------------|
| ad4f2a8c99 | George Joseph | debugging: Add enough to choke a mule |
|------------+---------------+-------------------------------------------|
| 7eaae4e7b6 | 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-18.0.0-summary.html | 1162 ----
asterisk-18.0.0-summary.txt | 2873 ----------
b/.version | 2
b/CHANGES | 30
b/ChangeLog | 742 ++
b/addons/ooh323c/src/ooq931.c | 2
b/apps/app_confbridge.c | 3
b/apps/app_directory.c | 2
b/apps/app_voicemail.c | 5
b/apps/confbridge/conf_config_parser.c | 21
b/apps/confbridge/include/confbridge.h | 2
b/asterisk-18.1.0-rc1-summary.html | 188
b/asterisk-18.1.0-rc1-summary.txt | 496 +
b/bridges/bridge_softmix.c | 153
b/channels/chan_audiosocket.c | 5
b/channels/chan_iax2.c | 2
b/channels/chan_pjsip.c | 94
b/channels/chan_sip.c | 13
b/configs/samples/confbridge.conf.sample | 7
b/configs/samples/features.conf.sample | 4
b/configs/samples/musiconhold.conf.sample | 4
b/configs/samples/res_curl.conf.sample | 1
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/contrib/scripts/sip_to_pjsip/astconfigparser.py | 43
b/funcs/func_curl.c | 48
b/funcs/func_odbc.c | 2
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/logger_category.h | 178
b/include/asterisk/pbx.h | 8
b/include/asterisk/res_pjsip.h | 94
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/res_stir_shaken.h | 3
b/include/asterisk/rtp_engine.h | 79
b/include/asterisk/sched.h | 5
b/include/asterisk/stream.h | 4
b/include/asterisk/stun.h | 25
b/main/bridge.c | 44
b/main/bridge_channel.c | 20
b/main/channel.c | 14
b/main/cli.c | 51
b/main/dsp.c | 4
b/main/format_cache.c | 21
b/main/format_cap.c | 2
b/main/indications.c | 6
b/main/logger.c | 5
b/main/logger_category.c | 324 +
b/main/pbx.c | 12
b/main/rtp_engine.c | 68
b/main/say.c | 20
b/main/stream.c | 30
b/main/stun.c | 61
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_audiosocket.c | 3
b/res/res_musiconhold.c | 24
b/res/res_parking.c | 1
b/res/res_pjsip.c | 56
b/res/res_pjsip/config_transport.c | 18
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_scheduler.c | 180
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_outbound_registration.c | 286
b/res/res_pjsip_pubsub.c | 10
b/res/res_pjsip_sdp_rtp.c | 26
b/res/res_pjsip_session.c | 1977 +++++-
b/res/res_pjsip_stir_shaken.c | 1
b/res/res_rtp_asterisk.c | 403 -
b/res/res_stasis.c | 31
b/res/res_stir_shaken.c | 4
b/res/res_stir_shaken/stir_shaken.c | 3
b/res/stasis/stasis_bridge.c | 2
84 files changed, 5520 insertions(+), 4976 deletions(-)

@ -0,0 +1,17 @@
<!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-18.1.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.1.1</h3><h3 align="center">Date: 2020-12-22</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="#open_issues">Open 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 has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
<li><a href="http://downloads.asterisk.org/pub/security/AST-2020-003,AST-2020-004.html">AST-2020-003,AST-2020-004</a></li>
</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-18.1.0.</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%">1 Torrey Searle <tsearle@voxbone.com><br/></td><td width="33%"><td width="33%">1 Mikhail Ivanov <mivanov@lanta-net.ru><br/>1 Torrey Searle <tsearle@gmail.com><br/></td></tr>
</table><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>Security</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29219">ASTERISK-29219</a>: res_pjsip_diversion: Crash if Tel URI contains History-Info<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5a48b86c9baacc99263ff21d387cc4708d7ea72">[e5a48b86c9]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
</ul><br><h3>Bug</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5a48b86c9baacc99263ff21d387cc4708d7ea72">[e5a48b86c9]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5a48b86c9baacc99263ff21d387cc4708d7ea72">[e5a48b86c9]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</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>

@ -0,0 +1,101 @@
Release Summary
asterisk-18.1.1
Date: 2020-12-22
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Open Issues
4. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release has been made to address one or more security vulnerabilities
that have been identified. A security advisory document has been published
for each vulnerability that includes additional information. Users of
versions of Asterisk that are affected are strongly encouraged to review
the advisories and determine what action they should take to protect their
systems from these issues.
Security Advisories:
* AST-2020-003,AST-2020-004
The data in this summary reflects changes that have been made since the
previous release, asterisk-18.1.0.
----------------------------------------------------------------------
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
1 Torrey Searle 1 Mikhail Ivanov
1 Torrey Searle
----------------------------------------------------------------------
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.
Security
Category: Resources/res_pjsip_diversion
ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains
History-Info
Reported by: Torrey Searle
* [e5a48b86c9] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
Bug
Category: Resources/res_pjsip_diversion
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [e5a48b86c9] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
Category: pjproject/pjsip
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [e5a48b86c9] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
----------------------------------------------------------------------
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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