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Asterisk Development Team
c11dc10be1 Update for 18.12.0 2022-05-12 06:50:59 -05:00
Asterisk Development Team
4c4d11eb6c Update for 18.12.0-rc1 2022-05-05 09:28:18 -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-18.12.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.12.0</h3><h3 align="center">Date: 2022-05-12</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-18.11.0.</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%">22 Naveen Albert <asterisk@phreaknet.org><br/>6 Joshua C. Colp <jcolp@sangoma.com><br/>5 Sean Bright <sean.bright@gmail.com><br/>4 Asterisk Development Team <asteriskteam@digium.com><br/>4 Ben Ford <bford@digium.com><br/>3 Mark Petersen <bugs.digium.com@zombie.dk><br/>2 Kevin Harwell <kharwell@sangoma.com><br/>2 Philip Prindeville <philipp@redfish-solutions.com><br/>2 George Joseph <gjoseph@digium.com><br/>2 Maximilian Fridrich <m.fridrich@commend.com><br/>1 Birger Harzenetter (license 5870)<br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Hugh McMaster <hugh.mcmaster@outlook.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Yury Kirsanov <y.kirsanov@gmail.com><br/>1 Marcel Wagner <mwagner@sipgate.de><br/></td><td width="33%"><td width="33%">18 N A <mail@interlinked.x10host.com><br/>3 Mark Petersen <asterisk.org@zombie.dk><br/>2 Rusty Newton <rnewton@digium.com><br/>2 Michael Auracher <m.auracher@commend.com><br/>2 George Joseph <gjoseph@digium.com><br/>2 Michael Auracher<br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Andre Heider <a.heider@gmail.com><br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Daniel Bonazzi <dbonazzi@arbeitsoftware.com><br/>1 Claude Diderich <claude.diderich@yahoo.com><br/>1 Scott Griepentrog <sgriepentrog@digium.com><br/>1 Arix <arix@xmail.re><br/>1 Stefan Ruijsenaars<br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 LA <learbia@gmail.com><br/>1 Josh Hogan <josh@vxt.co.nz><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Marcel Wagner <mwagner@sipgate.de><br/>1 INVADE International Ltd. <support@invade.net><br/>1 Leandro Dardini <ldardini@gmail.com><br/>1 Yury Kirsanov <y.kirsanov@gmail.com><br/>1 Clint Ruoho <clint@ruoho.org><br/>1 Jim Van Meggelen <jim.vanmeggelen@clearlycore.com><br/>1 Stefan Ruijsenaars <stefanr@wave.com><br/>1 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>1 Hugh McMaster <hugh.mcmaster@outlook.com><br/>1 Jonathan Harris <lardconcepts@gmail.com><br/>1 David Herselman <bbs2web@hotmail.com><br/>1 Dmitriy Serov <serov.d.p@gmail.com><br/>1 Philip Prindeville <philipp@redfish-solutions.com><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Jasper Hafkenscheid <jasper.hafkenscheid@wearespindle.com><br/>1 Kevin Harwell <kharwell@digium.com><br/>1 Josh Alberts <jmana9@gmail.com><br/>1 Sebastian Gutierrez <scgm11@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: Functions/func_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29838">ASTERISK-29838</a>: ${SQL_ESC()} not correctly escaping a terminating \<br/>Reported by: Leandro Dardini<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39cd09c246ab66f6e6e56f7543230247071ea72b">[39cd09c246]</a> Joshua C. Colp -- func_odbc: Add SQL_ESC_BACKSLASHES dialplan function.</li>
</ul><br><h4>Category: Resources/res_stir_shaken</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29476">ASTERISK-29476</a>: res_stir_shaken: Blind SSRF vulnerabilities<br/>Reported by: Clint Ruoho<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=11accf8064d8ea86ddba50b1065b1d7ade0cbd0c">[11accf8064]</a> Ben Ford -- AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29872">ASTERISK-29872</a>: res_stir_shaken: Resource exhaustion with large files<br/>Reported by: Benjamin Keith Ford<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=33091c2659a4cb28ea616c152018a574f951e881">[33091c2659]</a> Ben Ford -- AST-2022-001 - res_stir_shaken/curl: Limit file size and check start.</li>
</ul><br><h3>New Feature</h3><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29931">ASTERISK-29931</a>: Option to allow a user to not hear the join sound on enter but everyone else can<br/>Reported by: Michael Cargile<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=216a55408ea8c667956b030946d8bc863f1e2cda">[216a55408e]</a> Michael Cargile -- apps/confbridge: Added hear_own_join_sound option to control who hears sound_join</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29876">ASTERISK-29876</a>: app_queue: Add music on hold option<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b7edc08e33f7ffb832d32a54825f0b699098d0ef">[b7edc08e33]</a> Naveen Albert -- app_queue: Add music on hold option to Queue.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29941">ASTERISK-29941</a>: chan_pjsip: Add ability to send flash events<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bc6d42a279045e92488f16372f2d4b417b6c7ef">[8bc6d42a27]</a> Naveen Albert -- chan_pjsip: Add ability to send flash events.</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29820">ASTERISK-29820</a>: cli: Add command to evaluate a function<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3686a97d799f2aa60225ea04522c98c9e5114bba">[3686a97d79]</a> Naveen Albert -- cli: Add command to evaluate dialplan functions.</li>
</ul><br><h4>Category: Functions/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29486">ASTERISK-29486</a>: Hint-like extension value lookup function without device state<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79689d9df8c0426604dc4156521ed82aabd113cb">[79689d9df8]</a> Naveen Albert -- func_evalexten: Extension evaluation function.</li>
</ul><br><h4>Category: Functions/func_db</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29968">ASTERISK-29968</a>: func_db: Add a function to return cardinality of keys at prefix<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ce00f8758d61c2e05578311a5919f6c7edd7092b">[ce00f8758d]</a> Naveen Albert -- func_db: Add function to return cardinality at prefix</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30002">ASTERISK-30002</a>: app_meetme: Don't erroneously set global variables when channel is NULL<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff70b2aac6379e89ec3d74deff7eb3c6651eff6d">[ff70b2aac6]</a> Naveen Albert -- app_meetme: Don't erroneously set global variables.</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29253">ASTERISK-29253</a>: Incorrect bridging on transfer<br/>Reported by: Yury Kirsanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ac08fdcf8d311d5cfedccf5b54b851a0bad47b0">[6ac08fdcf8]</a> Yury Kirsanov -- bridge_simple.c: Unhold channels on join simple bridge.</li>
</ul><br><h4>Category: CDR/cdr_adaptive_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30023">ASTERISK-30023</a>: cdr_adaptive_odbc: does not support DATETIME database columns<br/>Reported by: Gregory Massel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec8ab44b7f207640ffa920b28efb2532c9f146ce">[ec8ab44b7f]</a> Joshua C. Colp -- cdr_adaptive_odbc: Add support for SQL_DATETIME field type.</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28518">ASTERISK-28518</a>: chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold<br/>Reported by: Josh Alberts<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a4f04666b5b54f3d633273fe46d1b78231877486">[a4f04666b5]</a> Naveen Albert -- chan_dahdi: Don't allow MWI FSK if channel not idle.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29990">ASTERISK-29990</a>: chan_dahdi: adding ring cadences is not idempotent on dahdi restart<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb53ad567107169d9cf802f4e34a69a1fc674f3c">[cb53ad5671]</a> Naveen Albert -- chan_dahdi: Don't append cadences on dahdi restart.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29994">ASTERISK-29994</a>: chan_dahdi: Round robin array size is too small for max number of groups<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd15cd049f2623e725eee601df0ae20071e54af3">[dd15cd049f]</a> Naveen Albert -- chan_dahdi: Fix insufficient array size for round robin.</li>
</ul><br><h4>Category: Channels/chan_iax2</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30007">ASTERISK-30007</a>: chan_iax2: Prevent crashes due to attempted encryption with missing secrets<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9dc321cbcb21a243b9e6eed48c0d8ee8aa24c689">[9dc321cbcb]</a> Naveen Albert -- chan_iax2: Prevent crash if dialing RSA-only call without outkey.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29895">ASTERISK-29895</a>: chan_iax2: Fix misaligned spacing in iax2 show netstats printout<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d9e55250ddd47bad36252ccd27c007cc5ec929f6">[d9e55250dd]</a> Birger Harzenetter -- chan_iax2: Fix spacing in netstats command</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29048">ASTERISK-29048</a>: chan_iax2: "iax2 show registry" shows host for perceived<br/>Reported by: David Herselman<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97c499ee343ee3680d5aeba49a29482f4c283415">[97c499ee34]</a> Naveen Albert -- chan_iax2: Fix perceived showing host address.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29842">ASTERISK-29842</a>: Do not change 180 Ringing to 183 Progress even if early_media already enabled<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16e59db5142034245d9ceb68fa8a1775f8353a7c">[16e59db514]</a> Mark Petersen -- chan_pjsip: add allow_sending_180_after_183 option</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30006">ASTERISK-30006</a>: res_pjsip: UDP transport does not work when async_operations is greater than 1<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09e8667fa5b421f7c39e29921e3f3a62456f7f2b">[09e8667fa5]</a> Joshua C. Colp -- res_pjsip: Always set async_operations to 1.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29843">ASTERISK-29843</a>: Session timers get removed on UPDATE<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb2102a9917b0c2ab86a8fccb73dce230f9a5cfc">[bb2102a991]</a> Mark Petersen -- chan_sip.c Session timers get removed on UPDATE</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29955">ASTERISK-29955</a>: chan_sip: SIP route header is missing on UPDATE<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f7e3d1609e391b5a3fac2ba509e58e4e68c28eb">[4f7e3d1609]</a> Mark Petersen -- chan_sip: SIP route header is missing on UPDATE</li>
</ul><br><h4>Category: Channels/chan_sip/Transfers</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29955">ASTERISK-29955</a>: chan_sip: SIP route header is missing on UPDATE<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f7e3d1609e391b5a3fac2ba509e58e4e68c28eb">[4f7e3d1609]</a> Mark Petersen -- chan_sip: SIP route header is missing on UPDATE</li>
</ul><br><h4>Category: Channels/chan_vpb</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30021">ASTERISK-30021</a>: ast_variable_list_replace_variable uses variable with new keyword<br/>Reported by: Jasper Hafkenscheid<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2587e58e05bf9f56aa20c86e5a31054330b10240">[2587e58e05]</a> Sean Bright -- config.h: Don't use C++ keywords as argument names.</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29986">ASTERISK-29986</a>: build: Asterisk 18.11.0 doesn't compile when wget isn't available<br/>Reported by: Stefan Ruijsenaars<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd704bbba516779bf671290789041b3de30b7fbe">[dd704bbba5]</a> George Joseph -- make_xml_documentation: Remove usage of get_sourceable_makeopts</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29988">ASTERISK-29988</a>: REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d3297d4f3e2cd944b8d7e2a69b0ba1da083f349">[2d3297d4f3]</a> George Joseph -- Makefile: Disable XML doc validation</li>
</ul><br><h4>Category: Core/FileFormatInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29943">ASTERISK-29943</a>: file.c: seeking to negative file offset is not prevented<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea02bc368518e3d2b952c9f27db48f414d667bc9">[ea02bc3685]</a> Naveen Albert -- file.c: Prevent formats from seeking negative offsets.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29948">ASTERISK-29948</a>: iostream: Infinite TCP timeout writing data<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae1373d12d0f774fffa3c965c542c9f0b137fe95">[ae1373d12d]</a> Joshua C. Colp -- manager: Terminate session on write error.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29674">ASTERISK-29674</a>: Adjust for 64bit time_t<br/>Reported by: Andre Heider<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f50e793665ea66b5cea7c612cc95ca27bf45afb8">[f50e793665]</a> Philip Prindeville -- time: add support for time64 libcs</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22246">ASTERISK-22246</a>: Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug)<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09e989f972e2583df4e9bf585c246c37322d8d2f">[09e989f972]</a> Naveen Albert -- asterisk.c: Warn of incompatibilities with remote console.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29928">ASTERISK-29928</a>: logging messages truncated when using MUSL runtime<br/>Reported by: Philip Prindeville<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=140c19c2067a5e2dcedfbb4dfa08c57758b822cb">[140c19c206]</a> Philip Prindeville -- logger: workaround woefully small BUFSIZ in MUSL</li>
</ul><br><h4>Category: Core/Netsock</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29948">ASTERISK-29948</a>: iostream: Infinite TCP timeout writing data<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae1373d12d0f774fffa3c965c542c9f0b137fe95">[ae1373d12d]</a> Joshua C. Colp -- manager: Terminate session on write error.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26719">ASTERISK-26719</a>: pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1)<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd69639a6b10e39848d6edd2a19c77f25800ef9b">[bd69639a6b]</a> Naveen Albert -- pbx.c: Warn if there are too many includes in a context.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29939">ASTERISK-29939</a>: agi: Fix xmldoc bug with set music<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc129b6951c6b207e7c1a686d72149b7a6460715">[dc129b6951]</a> Naveen Albert -- res_agi: Fix xmldocs bug with set music.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28891">ASTERISK-28891</a>: documentation: AGICommand_set+music documentation arguments displayed incorreclty<br/>Reported by: Jonathan Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc129b6951c6b207e7c1a686d72149b7a6460715">[dc129b6951]</a> Naveen Albert -- res_agi: Fix xmldocs bug with set music.</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29728">ASTERISK-29728</a>: menuselect: Disabled by default modules that are enabled are always recompiled<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4c17c2044a8c78421c64b50da8ff1dbf368994b">[b4c17c2044]</a> Naveen Albert -- menuselect: Don't erroneously recompile modules.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22246">ASTERISK-22246</a>: Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug)<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09e989f972e2583df4e9bf585c246c37322d8d2f">[09e989f972]</a> Naveen Albert -- asterisk.c: Warn of incompatibilities with remote console.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26582">ASTERISK-26582</a>: Asterisk seems to ignore the "n" parameter for "disable console colorization"<br/>Reported by: Sebastian Gutierrez<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09e989f972e2583df4e9bf585c246c37322d8d2f">[09e989f972]</a> Naveen Albert -- asterisk.c: Warn of incompatibilities with remote console.</li>
</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29950">ASTERISK-29950</a>: SayNumber can handle '01' to '07', but not '08' or '09'<br/>Reported by: Jim Van Meggelen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=81a990b8d2431b9299ff77646847a92fbe651786">[81a990b8d2]</a> Sean Bright -- conversions.c: Specify that we only want to parse decimal numbers.</li>
</ul><br><h4>Category: Resources/res_ari_recordings</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29960">ASTERISK-29960</a>: ari: Retrieving stored recording can returns wrong file<br/>Reported by: Arix<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a7d83087bcfc1f06b5ff66d3be24874c8217623">[3a7d83087b]</a> Sean Bright -- stasis_recording: Perform a complete match on requested filename.</li>
</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29411">ASTERISK-29411</a>: Crash in pjsip_msg_find_hdr_by_name<br/>Reported by: LA<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec5b449bcf97729f8664086049c1c4d92564be4e">[ec5b449bcf]</a> Kevin Harwell -- res_pjsip_header_funcs: wrong pool used tdata headers</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29961">ASTERISK-29961</a>: RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=96a3ff9eddd6a1964624c924fabb7f3d59a562dc">[96a3ff9edd]</a> Alexei Gradinari -- res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26689">ASTERISK-26689</a>: res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82dbfe7783d5199a940fb76b795f1ee1a3d4c9e5">[82dbfe7783]</a> Boris P. Korzun -- res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29929">ASTERISK-29929</a>: res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions<br/>Reported by: Boris P. Korzun<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82dbfe7783d5199a940fb76b795f1ee1a3d4c9e5">[82dbfe7783]</a> Boris P. Korzun -- res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29655">ASTERISK-29655</a>: res_pjsip_session: No video to caller if no camera available<br/>Reported by: Michael Auracher<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e6991f95e6a70146a64407ab01bbbe45f93745f">[1e6991f95e]</a> Maximilian Fridrich -- core_unreal: Flip stream direction of second channel.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37829b4461dadf798611af25de0d58f55eedbfa4">[37829b4461]</a> Maximilian Fridrich -- app_dial: Flip stream direction of outgoing channel.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29638">ASTERISK-29638</a>: res_pjsip_session: No video after early media<br/>Reported by: Michael Auracher<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e6991f95e6a70146a64407ab01bbbe45f93745f">[1e6991f95e]</a> Maximilian Fridrich -- core_unreal: Flip stream direction of second channel.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37829b4461dadf798611af25de0d58f55eedbfa4">[37829b4461]</a> Maximilian Fridrich -- app_dial: Flip stream direction of outgoing channel.</li>
</ul><br><h4>Category: Resources/res_stir_shaken</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30024">ASTERISK-30024</a>: Failed to sign STIR/SHAKEN payload with functionality not enabled<br/>Reported by: Claude Diderich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40f4268f2df3ac4d416e06bf825b2ea954a42075">[40f4268f2d]</a> Ben Ford -- res_pjsip_stir_shaken.c: Fix enabled when not configured.</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30015">ASTERISK-30015</a>: pjsip / WebRTC: Chrome creating large number of SDP attributes<br/>Reported by: Josh Hogan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=850021061178dbcf117f7bc1a0bf711838aa1efb">[8500210611]</a> Joshua C. Colp -- pjsip: Increase maximum number of format attributes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29535">ASTERISK-29535</a>: Segmentation fault in libasteriskpj.so.2<br/>Reported by: Daniel Bonazzi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec5b449bcf97729f8664086049c1c4d92564be4e">[ec5b449bcf]</a> Kevin Harwell -- res_pjsip_header_funcs: wrong pool used tdata headers</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29951">ASTERISK-29951</a>: app_mf, app_sf: Return -1 on hangup<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9024bb989b2776014213c4e7577f9ba06e208403">[9024bb989b]</a> Naveen Albert -- app_mf, app_sf: Return -1 if channel hangs up.</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25716">ASTERISK-25716</a>: Documentation: Document explanations and examples for possible values of DIALSTATUS<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a66b6647b296aac17713743541e54128bc761336">[a66b6647b2]</a> Naveen Albert -- app_dial: Document DIALSTATUS return values.</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29954">ASTERISK-29954</a>: app_meetme: Emit warning if conference not found<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b2d5bd4cb8a76d01e0509f244fcb121783e43544">[b2d5bd4cb8]</a> Naveen Albert -- app_meetme: Emit warning if conference not found.</li>
</ul><br><h4>Category: Codecs/codec_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29980">ASTERISK-29980</a>: build: External binary modules don't use https<br/>Reported by: INVADE International Ltd.<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b636f376603bcd98fba44bb9f488337613ce40d">[2b636f3766]</a> Sean Bright -- download_externals: Use HTTPS for downloads</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29970">ASTERISK-29970</a>: Use pkg-config to find libxml2 headers and libraries<br/>Reported by: Hugh McMaster<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b678624f0416388a41c9047f1bfcf9658403d3ac">[b678624f04]</a> Hugh McMaster -- configure.ac: Use pkg-config to detect libxml2</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29980">ASTERISK-29980</a>: build: External binary modules don't use https<br/>Reported by: INVADE International Ltd.<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b636f376603bcd98fba44bb9f488337613ce40d">[2b636f3766]</a> Sean Bright -- download_externals: Use HTTPS for downloads</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24827">ASTERISK-24827</a>: Missing documentation for chan_dahdi dial string ring cadences<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=085f33b7a363ffaaa749f5380580787895a846fd">[085f33b7a3]</a> Naveen Albert -- chan_dahdi: Document dial resource options.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29940">ASTERISK-29940</a>: general: Add since tags to xmldocs<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4aac359d79e1a59a83fd5b1b0e2ec06defa13218">[4aac359d79]</a> Naveen Albert -- documentation: Adds versioning information.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29976">ASTERISK-29976</a>: Should Readme include information about install_prereq script?<br/>Reported by: Marcel Wagner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a893fdd90110562cbe31428909d0043a1699a802">[a893fdd901]</a> Marcel Wagner -- documentation: Add information on running install_prereq script in readme</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25716">ASTERISK-25716</a>: Documentation: Document explanations and examples for possible values of DIALSTATUS<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a66b6647b296aac17713743541e54128bc761336">[a66b6647b2]</a> Naveen Albert -- app_dial: Document DIALSTATUS return values.</li>
</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29967">ASTERISK-29967</a>: pbx_builtins: Add missing documentation<br/>Reported by: N A<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b407511f024eb82ada81bc6083dd98e17e281167">[b407511f02]</a> Naveen Albert -- pbx_builtins: Add missing options documentation</li>
</ul><br><h4>Category: Resources/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29726">ASTERISK-29726</a>: Add Asterisk External Application Protocol (AEAP) implementation<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fb8667908a688a9b6568c7a51e5e20ad3b2d6d7">[2fb8667908]</a> Kevin Harwell -- res_aeap & res_speech_aeap: Add Asterisk External Application Protocol</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29351">ASTERISK-29351</a>: Qualify pjproject 2.12 for Asterisk<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5e02f783d66c6ea10001934c9da20b11fa2effc">[e5e02f783d]</a> Joshua C. Colp -- pjproject: Update bundled to 2.12 release.</li>
</ul><br><h4>Category: Resources/res_speech/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29726">ASTERISK-29726</a>: Add Asterisk External Application Protocol (AEAP) implementation<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fb8667908a688a9b6568c7a51e5e20ad3b2d6d7">[2fb8667908]</a> Kevin Harwell -- res_aeap & res_speech_aeap: Add Asterisk External Application Protocol</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=4c4d11eb6c153301894bf469594bd7a7c444ee27">4c4d11eb6c</a></td><td>Asterisk Development Team</td><td>Update for 18.12.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=efca7f4e8d932f9a1662d54cc53207edbe3106cc">efca7f4e8d</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.12.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62f8e157fb3c693bb90e17a49afa6a7257e56485">62f8e157fb</a></td><td>Ben Ford</td><td>res_aeap: Add basic config skeleton and CLI commands.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=801317ae05ccbaea29183fa2c3bba5a5844ceda6">801317ae05</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.11.2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f325cb3d13b424d06b1b47d20cfc7cfbebca9a6a">f325cb3d13</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.11.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=777e9fde6748bc83b2a067a9dcf5cdb14a62f60e">777e9fde67</a></td><td>Sean Bright</td><td>openssl: Supress deprecation warnings from OpenSSL 3.0</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-18.11.0-summary.html | 147
asterisk-18.11.0-summary.txt | 381
b/.version | 2
b/CHANGES | 76
b/ChangeLog | 910
b/Makefile | 9
b/README.md | 5
b/UPGRADE.txt | 12
b/apps/app_confbridge.c | 22
b/apps/app_dial.c | 55
b/apps/app_dtmfstore.c | 5
b/apps/app_meetme.c | 18
b/apps/app_mf.c | 29
b/apps/app_queue.c | 38
b/apps/app_sendtext.c | 5
b/apps/app_sf.c | 39
b/apps/app_waitforcond.c | 5
b/apps/confbridge/conf_config_parser.c | 7
b/apps/confbridge/include/confbridge.h | 1
b/asterisk-18.12.0-rc1-summary.html | 301
b/asterisk-18.12.0-rc1-summary.txt | 715
b/bridges/bridge_simple.c | 21
b/build_tools/download_externals | 2
b/build_tools/make_xml_documentation | 42
b/cdr/cdr_adaptive_odbc.c | 1
b/channels/chan_dahdi.c | 59
b/channels/chan_iax2.c | 34
b/channels/chan_pjsip.c | 38
b/channels/chan_sip.c | 14
b/channels/pjsip/dialplan_functions.c | 16
b/codecs/codecs.xml | 10
b/configs/samples/aeap.conf.sample | 21
b/configs/samples/chan_dahdi.conf.sample | 5
b/configs/samples/confbridge.conf.sample | 6
b/configs/samples/func_odbc.conf.sample | 4
b/configs/samples/pjsip.conf.sample | 12
b/configs/samples/queues.conf.sample | 3
b/configs/samples/stir_shaken.conf.sample | 18
b/configure |18310 +++++-----
b/configure.ac | 1
b/contrib/ast-db-manage/config/versions/0bee61aa9425_allow_180_ringing_with_sdp.py | 36
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/doc/asterisk.8 | 4
b/funcs/func_channel.c | 5
b/funcs/func_db.c | 72
b/funcs/func_env.c | 10
b/funcs/func_evalexten.c | 147
b/funcs/func_frame_drop.c | 5
b/funcs/func_json.c | 5
b/funcs/func_math.c | 15
b/funcs/func_odbc.c | 39
b/funcs/func_sayfiles.c | 5
b/funcs/func_scramble.c | 5
b/funcs/func_strings.c | 5
b/funcs/func_talkdetect.c | 3
b/include/asterisk/config.h | 13
b/include/asterisk/pbx.h | 19
b/include/asterisk/res_aeap.h | 370
b/include/asterisk/res_aeap_message.h | 374
b/include/asterisk/res_pjsip.h | 9
b/include/asterisk/res_stir_shaken.h | 54
b/include/asterisk/speech.h | 6
b/include/asterisk/time.h | 20
b/main/Makefile | 1
b/main/asterisk.c | 53
b/main/conversions.c | 4
b/main/core_unreal.c | 31
b/main/file.c | 6
b/main/logger.c | 15
b/main/manager.c | 8
b/main/pbx.c | 79
b/main/pbx_builtins.c | 7
b/main/pbx_variables.c | 54
b/main/time.c | 29
b/menuselect/configure | 3730 +-
b/menuselect/configure.ac | 2
b/menuselect/menuselect.c | 70
b/res/Makefile | 1
b/res/res.xml | 2
b/res/res_aeap.c | 404
b/res/res_aeap.exports.in | 7
b/res/res_aeap/aeap.c | 501
b/res/res_aeap/general.c | 58
b/res/res_aeap/general.h | 41
b/res/res_aeap/logger.h | 60
b/res/res_aeap/message.c | 270
b/res/res_aeap/message_json.c | 191
b/res/res_aeap/transaction.c | 284
b/res/res_aeap/transaction.h | 123
b/res/res_aeap/transport.c | 156
b/res/res_aeap/transport.h | 209
b/res/res_aeap/transport_websocket.c | 249
b/res/res_aeap/transport_websocket.h | 34
b/res/res_agi.c | 10
b/res/res_calendar_caldav.c | 4
b/res/res_calendar_icalendar.c | 4
b/res/res_crypto.c | 2
b/res/res_http_media_cache.c | 7
b/res/res_odbc.c | 4
b/res/res_pjsip/config_global.c | 21
b/res/res_pjsip/config_transport.c | 6
b/res/res_pjsip/location.c | 5
b/res/res_pjsip/pjsip_config.xml | 20
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_options.c | 4
b/res/res_pjsip_header_funcs.c | 8
b/res/res_pjsip_history.c | 25
b/res/res_pjsip_pubsub.c | 19
b/res/res_pjsip_registrar.c | 5
b/res/res_pjsip_sdp_rtp.c | 48
b/res/res_pjsip_stir_shaken.c | 24
b/res/res_rtp_asterisk.c | 1
b/res/res_speech.c | 36
b/res/res_speech_aeap.c | 731
b/res/res_stir_shaken.c | 96
b/res/res_stir_shaken/curl.c | 177
b/res/res_stir_shaken/curl.h | 5
b/res/res_stir_shaken/profile.c | 241
b/res/res_stir_shaken/profile.h | 39
b/res/res_stir_shaken/profile_private.h | 40
b/res/res_stir_shaken/stir_shaken.c | 29
b/res/res_stir_shaken/stir_shaken.h | 7
b/res/res_tonedetect.c | 15
b/res/stasis_recording/stored.c | 6
b/tests/test_aeap.c | 252
b/tests/test_aeap_speech.c | 287
b/tests/test_aeap_transaction.c | 179
b/tests/test_aeap_transport.c | 249
b/tests/test_conversions.c | 12
b/third-party/pjproject/patches/0000-configure-ssl-library-path.patch | 29
b/third-party/pjproject/patches/0000-remove-third-party.patch | 33
b/third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch | 413
build_tools/get_sourceable_makeopts | 54
third-party/pjproject/patches/0000-set_apps_initial_log_level.patch | 53
third-party/pjproject/patches/0000-solaris.patch | 135
third-party/pjproject/patches/0011-sip_inv_patch.patch | 39
third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch | 39
third-party/pjproject/patches/0050-fix-race-parallel-build.patch | 72
third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch | 28
third-party/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch | 37
third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 33
third-party/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch | 212
third-party/pjproject/patches/0100-fix-double-stun-free.patch | 82
third-party/pjproject/patches/0110-tls-parent-listener-destroyed.patch | 166
third-party/pjproject/patches/0111-ssl-premature-destroy.patch | 136
third-party/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch | 32
third-party/pjproject/patches/0130-sip_inv-Additional-multipart-support-2919-2920.patch | 661
third-party/pjproject/patches/0140-Fix-incorrect-unescaping-of-tokens-during-parsing-29.patch | 123
third-party/pjproject/patches/0150-Create-generic-pjsip_hdr_find-functions.patch | 176
third-party/pjproject/patches/0160-Additional-multipart-improvements.patch | 644
third-party/pjproject/patches/0170-stun-integer-underflow.patch | 26
third-party/pjproject/patches/0171-dialog-set-free.patch | 114
third-party/pjproject/patches/0172-prevent-multipart-oob.patch | 22
154 files changed, 21655 insertions(+), 13634 deletions(-)</pre><br></html>

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Release Summary
asterisk-18.12.0
Date: 2022-05-12
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
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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-18.11.0.
----------------------------------------------------------------------
Contributors
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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
22 Naveen Albert 18 N A
6 Joshua C. Colp 3 Mark Petersen
5 Sean Bright 2 Rusty Newton
4 Asterisk Development Team 2 Michael Auracher
4 Ben Ford 2 George Joseph
3 Mark Petersen 2 Michael Auracher
2 Kevin Harwell 1 Michael Cargile
2 Philip Prindeville 1 Andre Heider
2 George Joseph 1 Alexei Gradinari
2 Maximilian Fridrich 1 Daniel Bonazzi
1 Birger Harzenetter (license 5870) 1 Claude Diderich
1 Alexei Gradinari 1 Scott Griepentrog
1 Michael Cargile 1 Arix
1 Hugh McMaster 1 Stefan Ruijsenaars
1 Boris P. Korzun 1 Benjamin Keith Ford
1 Yury Kirsanov 1 LA
1 Marcel Wagner 1 Josh Hogan
1 Ross Beer
1 Boris P. Korzun
1 Marcel Wagner
1 INVADE International Ltd.
1 Leandro Dardini
1 Yury Kirsanov
1 Clint Ruoho
1 Jim Van Meggelen
1 Stefan Ruijsenaars
1 Tzafrir Cohen
1 Hugh McMaster
1 Jonathan Harris
1 David Herselman
1 Dmitriy Serov
1 Philip Prindeville
1 Gregory Massel
1 Jasper Hafkenscheid
1 Kevin Harwell
1 Josh Alberts
1 Sebastian Gutierrez
----------------------------------------------------------------------
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: Functions/func_odbc
ASTERISK-29838: ${SQL_ESC()} not correctly escaping a terminating \
Reported by: Leandro Dardini
* [39cd09c246] Joshua C. Colp -- func_odbc: Add SQL_ESC_BACKSLASHES
dialplan function.
Category: Resources/res_stir_shaken
ASTERISK-29476: res_stir_shaken: Blind SSRF vulnerabilities
Reported by: Clint Ruoho
* [11accf8064] Ben Ford -- AST-2022-002 - res_stir_shaken/curl: Add ACL
checks for Identity header.
ASTERISK-29872: res_stir_shaken: Resource exhaustion with large files
Reported by: Benjamin Keith Ford
* [33091c2659] Ben Ford -- AST-2022-001 - res_stir_shaken/curl: Limit
file size and check start.
New Feature
Category: Applications/app_confbridge
ASTERISK-29931: Option to allow a user to not hear the join sound on enter
but everyone else can
Reported by: Michael Cargile
* [216a55408e] Michael Cargile -- apps/confbridge: Added
hear_own_join_sound option to control who hears sound_join
Category: Applications/app_queue
ASTERISK-29876: app_queue: Add music on hold option
Reported by: N A
* [b7edc08e33] Naveen Albert -- app_queue: Add music on hold option to
Queue.
Category: Channels/chan_pjsip
ASTERISK-29941: chan_pjsip: Add ability to send flash events
Reported by: N A
* [8bc6d42a27] Naveen Albert -- chan_pjsip: Add ability to send flash
events.
Category: Functions/General
ASTERISK-29820: cli: Add command to evaluate a function
Reported by: N A
* [3686a97d79] Naveen Albert -- cli: Add command to evaluate dialplan
functions.
Category: Functions/NewFeature
ASTERISK-29486: Hint-like extension value lookup function without device
state
Reported by: N A
* [79689d9df8] Naveen Albert -- func_evalexten: Extension evaluation
function.
Category: Functions/func_db
ASTERISK-29968: func_db: Add a function to return cardinality of keys at
prefix
Reported by: N A
* [ce00f8758d] Naveen Albert -- func_db: Add function to return
cardinality at prefix
Bug
Category: Applications/app_meetme
ASTERISK-30002: app_meetme: Don't erroneously set global variables when
channel is NULL
Reported by: N A
* [ff70b2aac6] Naveen Albert -- app_meetme: Don't erroneously set global
variables.
Category: Bridges/bridge_simple
ASTERISK-29253: Incorrect bridging on transfer
Reported by: Yury Kirsanov
* [6ac08fdcf8] Yury Kirsanov -- bridge_simple.c: Unhold channels on join
simple bridge.
Category: CDR/cdr_adaptive_odbc
ASTERISK-30023: cdr_adaptive_odbc: does not support DATETIME database
columns
Reported by: Gregory Massel
* [ec8ab44b7f] Joshua C. Colp -- cdr_adaptive_odbc: Add support for
SQL_DATETIME field type.
Category: Channels/chan_dahdi
ASTERISK-28518: chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
Dahdi Call On Hold
Reported by: Josh Alberts
* [a4f04666b5] Naveen Albert -- chan_dahdi: Don't allow MWI FSK if
channel not idle.
ASTERISK-29990: chan_dahdi: adding ring cadences is not idempotent on
dahdi restart
Reported by: N A
* [cb53ad5671] Naveen Albert -- chan_dahdi: Don't append cadences on
dahdi restart.
ASTERISK-29994: chan_dahdi: Round robin array size is too small for max
number of groups
Reported by: N A
* [dd15cd049f] Naveen Albert -- chan_dahdi: Fix insufficient array size
for round robin.
Category: Channels/chan_iax2
ASTERISK-30007: chan_iax2: Prevent crashes due to attempted encryption
with missing secrets
Reported by: N A
* [9dc321cbcb] Naveen Albert -- chan_iax2: Prevent crash if dialing
RSA-only call without outkey.
ASTERISK-29895: chan_iax2: Fix misaligned spacing in iax2 show netstats
printout
Reported by: N A
* [d9e55250dd] Birger Harzenetter -- chan_iax2: Fix spacing in netstats
command
ASTERISK-29048: chan_iax2: "iax2 show registry" shows host for perceived
Reported by: David Herselman
* [97c499ee34] Naveen Albert -- chan_iax2: Fix perceived showing host
address.
Category: Channels/chan_pjsip
ASTERISK-29842: Do not change 180 Ringing to 183 Progress even if
early_media already enabled
Reported by: Mark Petersen
* [16e59db514] Mark Petersen -- chan_pjsip: add
allow_sending_180_after_183 option
ASTERISK-30006: res_pjsip: UDP transport does not work when
async_operations is greater than 1
Reported by: Ross Beer
* [09e8667fa5] Joshua C. Colp -- res_pjsip: Always set async_operations
to 1.
Category: Channels/chan_sip/General
ASTERISK-29843: Session timers get removed on UPDATE
Reported by: Mark Petersen
* [bb2102a991] Mark Petersen -- chan_sip.c Session timers get removed on
UPDATE
ASTERISK-29955: chan_sip: SIP route header is missing on UPDATE
Reported by: Mark Petersen
* [4f7e3d1609] Mark Petersen -- chan_sip: SIP route header is missing on
UPDATE
Category: Channels/chan_sip/Transfers
ASTERISK-29955: chan_sip: SIP route header is missing on UPDATE
Reported by: Mark Petersen
* [4f7e3d1609] Mark Petersen -- chan_sip: SIP route header is missing on
UPDATE
Category: Channels/chan_vpb
ASTERISK-30021: ast_variable_list_replace_variable uses variable with new
keyword
Reported by: Jasper Hafkenscheid
* [2587e58e05] Sean Bright -- config.h: Don't use C++ keywords as
argument names.
Category: Core/BuildSystem
ASTERISK-29986: build: Asterisk 18.11.0 doesn't compile when wget isn't
available
Reported by: Stefan Ruijsenaars
* [dd704bbba5] George Joseph -- make_xml_documentation: Remove usage of
get_sourceable_makeopts
ASTERISK-29988: REGRESSION: The build process is requiring xmllint or
xmlstarlet ro be installed when it shouldn't
Reported by: George Joseph
* [2d3297d4f3] George Joseph -- Makefile: Disable XML doc validation
Category: Core/FileFormatInterface
ASTERISK-29943: file.c: seeking to negative file offset is not prevented
Reported by: N A
* [ea02bc3685] Naveen Albert -- file.c: Prevent formats from seeking
negative offsets.
Category: Core/General
ASTERISK-29948: iostream: Infinite TCP timeout writing data
Reported by: N A
* [ae1373d12d] Joshua C. Colp -- manager: Terminate session on write
error.
ASTERISK-29674: Adjust for 64bit time_t
Reported by: Andre Heider
* [f50e793665] Philip Prindeville -- time: add support for time64 libcs
Category: Core/Logging
ASTERISK-22246: Asterisk's "T" flag is ignored when used with "r" or "R"
flags. (documentation bug)
Reported by: Rusty Newton
* [09e989f972] Naveen Albert -- asterisk.c: Warn of incompatibilities
with remote console.
ASTERISK-29928: logging messages truncated when using MUSL runtime
Reported by: Philip Prindeville
* [140c19c206] Philip Prindeville -- logger: workaround woefully small
BUFSIZ in MUSL
Category: Core/Netsock
ASTERISK-29948: iostream: Infinite TCP timeout writing data
Reported by: N A
* [ae1373d12d] Joshua C. Colp -- manager: Terminate session on write
error.
Category: Core/PBX
ASTERISK-26719: pbx: Only up to 127 includes in a dialplan context
(AST_PBX_MAX_STACK - 1)
Reported by: Tzafrir Cohen
* [bd69639a6b] Naveen Albert -- pbx.c: Warn if there are too many
includes in a context.
Category: Documentation
ASTERISK-29939: agi: Fix xmldoc bug with set music
Reported by: N A
* [dc129b6951] Naveen Albert -- res_agi: Fix xmldocs bug with set music.
ASTERISK-28891: documentation: AGICommand_set+music documentation
arguments displayed incorreclty
Reported by: Jonathan Harris
* [dc129b6951] Naveen Albert -- res_agi: Fix xmldocs bug with set music.
Category: General
ASTERISK-29728: menuselect: Disabled by default modules that are enabled
are always recompiled
Reported by: N A
* [b4c17c2044] Naveen Albert -- menuselect: Don't erroneously recompile
modules.
ASTERISK-22246: Asterisk's "T" flag is ignored when used with "r" or "R"
flags. (documentation bug)
Reported by: Rusty Newton
* [09e989f972] Naveen Albert -- asterisk.c: Warn of incompatibilities
with remote console.
ASTERISK-26582: Asterisk seems to ignore the "n" parameter for "disable
console colorization"
Reported by: Sebastian Gutierrez
* [09e989f972] Naveen Albert -- asterisk.c: Warn of incompatibilities
with remote console.
Category: PBX/General
ASTERISK-29950: SayNumber can handle '01' to '07', but not '08' or '09'
Reported by: Jim Van Meggelen
* [81a990b8d2] Sean Bright -- conversions.c: Specify that we only want
to parse decimal numbers.
Category: Resources/res_ari_recordings
ASTERISK-29960: ari: Retrieving stored recording can returns wrong file
Reported by: Arix
* [3a7d83087b] Sean Bright -- stasis_recording: Perform a complete match
on requested filename.
Category: Resources/res_pjsip_nat
ASTERISK-29411: Crash in pjsip_msg_find_hdr_by_name
Reported by: LA
* [ec5b449bcf] Kevin Harwell -- res_pjsip_header_funcs: wrong pool used
tdata headers
Category: Resources/res_pjsip_pubsub
ASTERISK-29961: RLS: domain part of 'uri' list attribute mismatch with
SUBSCRIBE request
Reported by: Alexei Gradinari
* [96a3ff9edd] Alexei Gradinari -- res_pjsip_pubsub: RLS 'uri' list
attribute mismatch with SUBSCRIBE request
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-26689: res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
channel for lack of RTP activity
Reported by: Dmitriy Serov
* [82dbfe7783] Boris P. Korzun -- res_pjsip_sdp_rtp: Improve detecting
of lack of RTP activity
ASTERISK-29929: res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
activity in one way sessions
Reported by: Boris P. Korzun
* [82dbfe7783] Boris P. Korzun -- res_pjsip_sdp_rtp: Improve detecting
of lack of RTP activity
Category: Resources/res_pjsip_session
ASTERISK-29655: res_pjsip_session: No video to caller if no camera
available
Reported by: Michael Auracher
* [1e6991f95e] Maximilian Fridrich -- core_unreal: Flip stream direction
of second channel.
* [37829b4461] Maximilian Fridrich -- app_dial: Flip stream direction of
outgoing channel.
ASTERISK-29638: res_pjsip_session: No video after early media
Reported by: Michael Auracher
* [1e6991f95e] Maximilian Fridrich -- core_unreal: Flip stream direction
of second channel.
* [37829b4461] Maximilian Fridrich -- app_dial: Flip stream direction of
outgoing channel.
Category: Resources/res_stir_shaken
ASTERISK-30024: Failed to sign STIR/SHAKEN payload with functionality not
enabled
Reported by: Claude Diderich
* [40f4268f2d] Ben Ford -- res_pjsip_stir_shaken.c: Fix enabled when not
configured.
Category: pjproject/pjsip
ASTERISK-30015: pjsip / WebRTC: Chrome creating large number of SDP
attributes
Reported by: Josh Hogan
* [8500210611] Joshua C. Colp -- pjsip: Increase maximum number of
format attributes.
ASTERISK-29535: Segmentation fault in libasteriskpj.so.2
Reported by: Daniel Bonazzi
* [ec5b449bcf] Kevin Harwell -- res_pjsip_header_funcs: wrong pool used
tdata headers
Improvement
Category: Applications/General
ASTERISK-29951: app_mf, app_sf: Return -1 on hangup
Reported by: N A
* [9024bb989b] Naveen Albert -- app_mf, app_sf: Return -1 if channel
hangs up.
Category: Applications/app_dial
ASTERISK-25716: Documentation: Document explanations and examples for
possible values of DIALSTATUS
Reported by: Rusty Newton
* [a66b6647b2] Naveen Albert -- app_dial: Document DIALSTATUS return
values.
Category: Applications/app_meetme
ASTERISK-29954: app_meetme: Emit warning if conference not found
Reported by: N A
* [b2d5bd4cb8] Naveen Albert -- app_meetme: Emit warning if conference
not found.
Category: Codecs/codec_opus
ASTERISK-29980: build: External binary modules don't use https
Reported by: INVADE International Ltd.
* [2b636f3766] Sean Bright -- download_externals: Use HTTPS for
downloads
Category: Core/BuildSystem
ASTERISK-29970: Use pkg-config to find libxml2 headers and libraries
Reported by: Hugh McMaster
* [b678624f04] Hugh McMaster -- configure.ac: Use pkg-config to detect
libxml2
ASTERISK-29980: build: External binary modules don't use https
Reported by: INVADE International Ltd.
* [2b636f3766] Sean Bright -- download_externals: Use HTTPS for
downloads
Category: Documentation
ASTERISK-24827: Missing documentation for chan_dahdi dial string ring
cadences
Reported by: Scott Griepentrog
* [085f33b7a3] Naveen Albert -- chan_dahdi: Document dial resource
options.
ASTERISK-29940: general: Add since tags to xmldocs
Reported by: N A
* [4aac359d79] Naveen Albert -- documentation: Adds versioning
information.
ASTERISK-29976: Should Readme include information about install_prereq
script?
Reported by: Marcel Wagner
* [a893fdd901] Marcel Wagner -- documentation: Add information on
running install_prereq script in readme
ASTERISK-25716: Documentation: Document explanations and examples for
possible values of DIALSTATUS
Reported by: Rusty Newton
* [a66b6647b2] Naveen Albert -- app_dial: Document DIALSTATUS return
values.
Category: PBX/General
ASTERISK-29967: pbx_builtins: Add missing documentation
Reported by: N A
* [b407511f02] Naveen Albert -- pbx_builtins: Add missing options
documentation
Category: Resources/NewFeature
ASTERISK-29726: Add Asterisk External Application Protocol (AEAP)
implementation
Reported by: Kevin Harwell
* [2fb8667908] Kevin Harwell -- res_aeap & res_speech_aeap: Add Asterisk
External Application Protocol
Category: Resources/res_pjsip
ASTERISK-29351: Qualify pjproject 2.12 for Asterisk
Reported by: George Joseph
* [e5e02f783d] Joshua C. Colp -- pjproject: Update bundled to 2.12
release.
Category: Resources/res_speech/NewFeature
ASTERISK-29726: Add Asterisk External Application Protocol (AEAP)
implementation
Reported by: Kevin Harwell
* [2fb8667908] Kevin Harwell -- res_aeap & res_speech_aeap: Add Asterisk
External Application Protocol
----------------------------------------------------------------------
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 |
|------------+----------------------+------------------------------------|
| 4c4d11eb6c | Asterisk Development | Update for 18.12.0-rc1 |
| | Team | |
|------------+----------------------+------------------------------------|
| efca7f4e8d | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 18.12.0 |
|------------+----------------------+------------------------------------|
| 62f8e157fb | Ben Ford | res_aeap: Add basic config |
| | | skeleton and CLI commands. |
|------------+----------------------+------------------------------------|
| 801317ae05 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 18.11.2 |
|------------+----------------------+------------------------------------|
| f325cb3d13 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 18.11.0 |
|------------+----------------------+------------------------------------|
| 777e9fde67 | Sean Bright | openssl: Supress deprecation |
| | | warnings from OpenSSL 3.0 |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-18.11.0-summary.html | 147
asterisk-18.11.0-summary.txt | 381
b/.version | 2
b/CHANGES | 76
b/ChangeLog | 910
b/Makefile | 9
b/README.md | 5
b/UPGRADE.txt | 12
b/apps/app_confbridge.c | 22
b/apps/app_dial.c | 55
b/apps/app_dtmfstore.c | 5
b/apps/app_meetme.c | 18
b/apps/app_mf.c | 29
b/apps/app_queue.c | 38
b/apps/app_sendtext.c | 5
b/apps/app_sf.c | 39
b/apps/app_waitforcond.c | 5
b/apps/confbridge/conf_config_parser.c | 7
b/apps/confbridge/include/confbridge.h | 1
b/asterisk-18.12.0-rc1-summary.html | 301
b/asterisk-18.12.0-rc1-summary.txt | 715
b/bridges/bridge_simple.c | 21
b/build_tools/download_externals | 2
b/build_tools/make_xml_documentation | 42
b/cdr/cdr_adaptive_odbc.c | 1
b/channels/chan_dahdi.c | 59
b/channels/chan_iax2.c | 34
b/channels/chan_pjsip.c | 38
b/channels/chan_sip.c | 14
b/channels/pjsip/dialplan_functions.c | 16
b/codecs/codecs.xml | 10
b/configs/samples/aeap.conf.sample | 21
b/configs/samples/chan_dahdi.conf.sample | 5
b/configs/samples/confbridge.conf.sample | 6
b/configs/samples/func_odbc.conf.sample | 4
b/configs/samples/pjsip.conf.sample | 12
b/configs/samples/queues.conf.sample | 3
b/configs/samples/stir_shaken.conf.sample | 18
b/configure |18310 +++++-----
b/configure.ac | 1
b/contrib/ast-db-manage/config/versions/0bee61aa9425_allow_180_ringing_with_sdp.py | 36
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/doc/asterisk.8 | 4
b/funcs/func_channel.c | 5
b/funcs/func_db.c | 72
b/funcs/func_env.c | 10
b/funcs/func_evalexten.c | 147
b/funcs/func_frame_drop.c | 5
b/funcs/func_json.c | 5
b/funcs/func_math.c | 15
b/funcs/func_odbc.c | 39
b/funcs/func_sayfiles.c | 5
b/funcs/func_scramble.c | 5
b/funcs/func_strings.c | 5
b/funcs/func_talkdetect.c | 3
b/include/asterisk/config.h | 13
b/include/asterisk/pbx.h | 19
b/include/asterisk/res_aeap.h | 370
b/include/asterisk/res_aeap_message.h | 374
b/include/asterisk/res_pjsip.h | 9
b/include/asterisk/res_stir_shaken.h | 54
b/include/asterisk/speech.h | 6
b/include/asterisk/time.h | 20
b/main/Makefile | 1
b/main/asterisk.c | 53
b/main/conversions.c | 4
b/main/core_unreal.c | 31
b/main/file.c | 6
b/main/logger.c | 15
b/main/manager.c | 8
b/main/pbx.c | 79
b/main/pbx_builtins.c | 7
b/main/pbx_variables.c | 54
b/main/time.c | 29
b/menuselect/configure | 3730 +-
b/menuselect/configure.ac | 2
b/menuselect/menuselect.c | 70
b/res/Makefile | 1
b/res/res.xml | 2
b/res/res_aeap.c | 404
b/res/res_aeap.exports.in | 7
b/res/res_aeap/aeap.c | 501
b/res/res_aeap/general.c | 58
b/res/res_aeap/general.h | 41
b/res/res_aeap/logger.h | 60
b/res/res_aeap/message.c | 270
b/res/res_aeap/message_json.c | 191
b/res/res_aeap/transaction.c | 284
b/res/res_aeap/transaction.h | 123
b/res/res_aeap/transport.c | 156
b/res/res_aeap/transport.h | 209
b/res/res_aeap/transport_websocket.c | 249
b/res/res_aeap/transport_websocket.h | 34
b/res/res_agi.c | 10
b/res/res_calendar_caldav.c | 4
b/res/res_calendar_icalendar.c | 4
b/res/res_crypto.c | 2
b/res/res_http_media_cache.c | 7
b/res/res_odbc.c | 4
b/res/res_pjsip/config_global.c | 21
b/res/res_pjsip/config_transport.c | 6
b/res/res_pjsip/location.c | 5
b/res/res_pjsip/pjsip_config.xml | 20
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_options.c | 4
b/res/res_pjsip_header_funcs.c | 8
b/res/res_pjsip_history.c | 25
b/res/res_pjsip_pubsub.c | 19
b/res/res_pjsip_registrar.c | 5
b/res/res_pjsip_sdp_rtp.c | 48
b/res/res_pjsip_stir_shaken.c | 24
b/res/res_rtp_asterisk.c | 1
b/res/res_speech.c | 36
b/res/res_speech_aeap.c | 731
b/res/res_stir_shaken.c | 96
b/res/res_stir_shaken/curl.c | 177
b/res/res_stir_shaken/curl.h | 5
b/res/res_stir_shaken/profile.c | 241
b/res/res_stir_shaken/profile.h | 39
b/res/res_stir_shaken/profile_private.h | 40
b/res/res_stir_shaken/stir_shaken.c | 29
b/res/res_stir_shaken/stir_shaken.h | 7
b/res/res_tonedetect.c | 15
b/res/stasis_recording/stored.c | 6
b/tests/test_aeap.c | 252
b/tests/test_aeap_speech.c | 287
b/tests/test_aeap_transaction.c | 179
b/tests/test_aeap_transport.c | 249
b/tests/test_conversions.c | 12
b/third-party/pjproject/patches/0000-configure-ssl-library-path.patch | 29
b/third-party/pjproject/patches/0000-remove-third-party.patch | 33
b/third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch | 413
build_tools/get_sourceable_makeopts | 54
third-party/pjproject/patches/0000-set_apps_initial_log_level.patch | 53
third-party/pjproject/patches/0000-solaris.patch | 135
third-party/pjproject/patches/0011-sip_inv_patch.patch | 39
third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch | 39
third-party/pjproject/patches/0050-fix-race-parallel-build.patch | 72
third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch | 28
third-party/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch | 37
third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 33
third-party/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch | 212
third-party/pjproject/patches/0100-fix-double-stun-free.patch | 82
third-party/pjproject/patches/0110-tls-parent-listener-destroyed.patch | 166
third-party/pjproject/patches/0111-ssl-premature-destroy.patch | 136
third-party/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch | 32
third-party/pjproject/patches/0130-sip_inv-Additional-multipart-support-2919-2920.patch | 661
third-party/pjproject/patches/0140-Fix-incorrect-unescaping-of-tokens-during-parsing-29.patch | 123
third-party/pjproject/patches/0150-Create-generic-pjsip_hdr_find-functions.patch | 176
third-party/pjproject/patches/0160-Additional-multipart-improvements.patch | 644
third-party/pjproject/patches/0170-stun-integer-underflow.patch | 26
third-party/pjproject/patches/0171-dialog-set-free.patch | 114
third-party/pjproject/patches/0172-prevent-multipart-oob.patch | 22
154 files changed, 21655 insertions(+), 13634 deletions(-)

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@@ -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';

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@@ -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';

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@@ -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

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@@ -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;