Update for 18.3.0

Update for 18.3.0-rc2
res_pjsip_session: Make reschedule_reinvite check for NULL topologies
2025-09-18 10:51:40 +00:00 · 2021-03-25 12:34:21 -05:00 · 2021-03-22 10:41:13 -05:00 · 2021-03-22 09:40:32 -05:00 · 2021-03-19 09:02:19 -06:00 · 2021-03-11 12:23:06 -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-18.3.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.3.0</h3><h3 align="center">Date: 2021-03-25</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-18.2.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">11 Alexander Traud <pabstraud@compuserve.com><br/>9 Joshua C. Colp <jcolp@sangoma.com><br/>8 Sean Bright <sean.bright@gmail.com><br/>6 Jaco Kroon <jaco@uls.co.za><br/>5 George Joseph <gjoseph@digium.com><br/>4 Ben Ford <bford@digium.com><br/>3 Kevin Harwell <kharwell@sangoma.com><br/>3 Asterisk Development Team <asteriskteam@digium.com><br/>3 Ivan Poddubnyi <ivan.poddubny@gmail.com><br/>3 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Salah Ahmed <sahmed@voxbone.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Holger Hans Peter Freyther <holger@moiji-mobile.com><br/>1 Nico Kooijman <nk@voclarion.nl><br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Torrey Searle <tsearle@voxbone.com><br/>1 Nick French <nickfrench@gmail.com><br/>1 Robert Cripps <rcripps@voxbone.com><br/>1 Sebastien Duthil <sduthil@wazo.community><br/>1 Mark Petersen <bugs.digium.com@zombie.dk><br/></td><td width="33%">1 Mark Petersen<br/></td><td width="33%">6 Alexander Traud <pabstraud@compuserve.com><br/>3 Boris P. Korzun <drtr0jan@yandex.ru><br/>3 Joshua C. Colp <jcolp@digium.com><br/>2 Matthias Hensler <mh@relaix.net><br/>2 Stefan Ruf <ruf.stefan@swm.de><br/>2 Sebastian Damm <sdamm@pascom.net><br/>2 Gregory Massel <greg@csurf.co.za><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Jacek Konieczny <jkonieczny@eggsoft.pl><br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Edvin Vidmar <edvinvidmar@hotmail.com><br/>1 Sébastien Duthil <sduthil@wazo.community><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 sungtae kim <pchero21@gmail.com><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Boolah  <boolah@mailvoid.net><br/>1 Nick French <nickfrench@gmail.com><br/>1 Salah Ahmed <txrubel@gmail.com><br/>1 Mauri de Souza Meneguzzo (3CPlus) <mauri.nunes@fluxoti.com><br/>1 N A<br/>1 N A <mail@interlinked.x10host.com><br/>1 Jacek Konieczny<br/>1 IAMJames_ <jamesys@gmail.com><br/>1 Mark Petersen <bugs.digium.com@zombie.dk><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Ivan Poddubny<br/>1 Vitezslav Novy <a1@vnovy.net><br/>1 Mark Petersen<br/>1 Michael Maier <m1278468@mailbox.org><br/>1 George Joseph <gjoseph@digium.com><br/>1 Alexander Traud<br/>1 Brian Paboojian <brian@nthonet.com><br/>1 Dan Cropp<br/>1 Robert Cripps <rcripps@voxbone.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: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29305">ASTERISK-29305</a>: ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash<br/>Reported by: Gregory Massel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77328142b439235d6423345603a0a59905e54c96">[77328142b4]</a> Ben Ford -- AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.</li>
+</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29260">ASTERISK-29260</a>: sRTP Replay Protection ignored; even tears down long calls<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=703158b9036ee278945be5dd9964405fb6c8b218">[703158b903]</a> Alexander Traud -- rtp:  Enable srtp replay protection</li>
+</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29227">ASTERISK-29227</a>: res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash<br/>Reported by: Ivan Poddubny<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2770cc58729398da402870302d5f56c034024a4a">[2770cc5872]</a> Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more than one histinfo to Supported</li>
+</ul><br><h3>Bug</h3><h4>Category: Applications/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29287">ASTERISK-29287</a>: app.h: C++ compatibility broken<br/>Reported by: Jean Aunis - Prescom<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=916d5d5e45656ae638352dda7119549e14184e10">[916d5d5e45]</a> Jaco Kroon -- app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS</li>
+</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29071">ASTERISK-29071</a>: app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs<br/>Reported by: Stefan Ruf<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7bda066bbfa558c8a2159a614b6792e4ed032a0">[f7bda066bb]</a> Joshua C. Colp -- channel: Fix crash in suppress API.</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b43b81d953a5a9e25bda84f81b45c2515c7eb4af">[b43b81d953]</a> Joshua C. Colp -- channel: Fix memory leak in suppress API.</li>
+</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29329">ASTERISK-29329</a>: app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events<br/>Reported by: N A<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94debe50858385e74657c56d607c34c1af02bc59">[94debe5085]</a> Sean Bright -- app_dial.c: Only send DTMF on first progress event.</li>
+</ul><br><h4>Category: Applications/app_page</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16799">ASTERISK-16799</a>: Callee declined when 'beep' audio file does not exist<br/>Reported by: IAMJames_<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6673c1b177d42601c8d4b0b3358785a646321df1">[6673c1b177]</a> Sean Bright -- app_page.c: Don't fail to Page if beep sound file is missing</li>
+</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28369">ASTERISK-28369</a>: app_queue: Member device state "invalid" when second call is ringing and hint is used<br/>Reported by: Boolah <ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=985d3e4940ccb92309fd45117501680f56ed1275">[985d3e4940]</a> Ivan Poddubnyi -- app_queue: Fix conversion of complex extension states into device states</li>
+</ul><br><h4>Category: Channels/chan_local</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29035">ASTERISK-29035</a>: chan_local: Multistream support breaks T.38 faxing<br/>Reported by: Matthias Hensler<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=47e9ce96ea306b22c6442a8b39006844b03fef8d">[47e9ce96ea]</a> Joshua C. Colp -- core_unreal: Fix deadlock with T.38 control frames.</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62e2dd484da4eb184056f6cc8d2e3b3e124e2dcc">[62e2dd484d]</a> Ben Ford -- core_unreal: Fix T.38 faxing when using local channels.</li>
+</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29280">ASTERISK-29280</a>: chan_sip: Allow peers without audio (text+video).<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45e48e387c5dfc6d719455777567d5aa580efff8">[45e48e387c]</a> Alexander Traud -- chan_sip: Allow [peer] without audio (text+video).</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29265">ASTERISK-29265</a>: chan_sip: Allow text+video media streams, again.<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87ad1138ffec1f87a765447999270b1bdacf535f">[87ad1138ff]</a> Alexander Traud -- chan_sip: Set up calls without audio (text+video), again.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29258">ASTERISK-29258</a>: chan_sip: Audio stream rejected, Other stream present: Invalid SDP.<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c154f3431183cdc8afce7c0ea6bff0ed44fe043">[4c154f3431]</a> Alexander Traud -- chan_sip: SDP: Reject audio streams correctly.</li>
+</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29071">ASTERISK-29071</a>: app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs<br/>Reported by: Stefan Ruf<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7bda066bbfa558c8a2159a614b6792e4ed032a0">[f7bda066bb]</a> Joshua C. Colp -- channel: Fix crash in suppress API.</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b43b81d953a5a9e25bda84f81b45c2515c7eb4af">[b43b81d953]</a> Joshua C. Colp -- channel: Fix memory leak in suppress API.</li>
+</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29259">ASTERISK-29259</a>: channel: Allow text+video media streams, again.<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f64ddf3db3c1b139d19dc6200f09136acd50108e">[f64ddf3db3]</a> Alexander Traud -- channel: Set up calls without audio (text+video), again.</li>
+</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29306">ASTERISK-29306</a>: strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition<br/>Reported by: Vitezslav Novy<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4cd7a7d0bf8b69a9fdbabaab5b9c4fa48e42cbb">[e4cd7a7d0b]</a> Sean Bright -- strings.h: ast_str_to_upper() and _to_lower() are not pure.</li>
+</ul><br><h4>Category: Core/Internationalization</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29297">ASTERISK-29297</a>: say: Y2021 problem – Asterisk cannot say year 2021 in Dutch<br/>Reported by: Jacek Konieczny<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b052ec965f68cb8d43a91c40e5ef8be8eaa2a9b">[7b052ec965]</a> Nico Kooijman -- main: With Dutch language year after 2020 is not spoken in say.c</li>
+</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24434">ASTERISK-24434</a>: Fix differing usage of assignment operators in modules.conf<br/>Reported by: Rusty Newton<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=30840846488f64664a2cb0fbaff51d6d3d5cb632">[3084084648]</a> Sean Bright -- modules.conf: Fix differing usage of assignment operators.</li>
+</ul><br><h4>Category: Resources/res_config_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29293">ASTERISK-29293</a>: res_config_pgsql: Limit realtime_pgsql() to return one (no more) record<br/>Reported by: Boris P. Korzun<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=beb579bc9909eaa0c2db4a3aa375565448d203be">[beb579bc99]</a> Boris P. Korzun -- res_config_pgsql: Limit realtime_pgsql() to return one (no more) record.</li>
+</ul><br><h4>Category: Resources/res_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29312">ASTERISK-29312</a>: res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters<br/>Reported by: Alexei Gradinari<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5e73d2121b80dbe8b381086f6c3fc1578dd9609">[d5e73d2121]</a> Alexei Gradinari -- res_fax: validate the remote/local Station ID for UTF-8 format</li>
+</ul><br><h4>Category: Resources/res_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29311">ASTERISK-29311</a>: res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit<br/>Reported by: Jaco Kroon<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ab53fce7a66957bd527b755664d97d947662445">[7ab53fce7a]</a> Jaco Kroon -- res_odbc_transaction: correctly initialise forcecommit value from DSN.</li>
+</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29196">ASTERISK-29196</a>: res_pjsip: Segmentation fault<br/>Reported by: Mauri de Souza Meneguzzo (3CPlus)<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=acb7ce4fe78c9b04baa182dc6cea4ffb047f10ea">[acb7ce4fe7]</a> Joshua C. Colp -- pjsip: Make modify_local_offer2 tolerate previous failed SDP.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29261">ASTERISK-29261</a>: res_pjsip: user=phone validation fail for isup numbers containing *#<br/>Reported by: Mark Petersen<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=176274caa4c4c85b6c7d8022b01e48bc59fe2843">[176274caa4]</a> Mark Petersen -- res/res_pjsip.c: allow user=phone when number contain *#</li>
+</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29235">ASTERISK-29235</a>: res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address<br/>Reported by: Brian Paboojian<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=976b1a1d7ac4dfca60e43e9f36f73510b22b23d6">[976b1a1d7a]</a> Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on REGISTER responses.</li>
+</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29315">ASTERISK-29315</a>: res_pjsip: re-registration gets stuck if setting initial auth credentials fails<br/>Reported by: Nick French<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dedfb334bdd4726867721958517576a91f9e81f1">[dedfb334bd]</a> Nick French -- res_pjsip: dont return early from registration if init auth fails</li>
+</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29313">ASTERISK-29313</a>: res_pjsip_refer:  Segfault in progress notify<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=15afabdf8e8dd65dadaa0d1111016a6ce94e1bdd">[15afabdf8e]</a> George Joseph -- res_pjsip_refer: Refactor progress locking and serialization</li>
+</ul><br><h4>Category: Resources/res_pjsip_registrar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29235">ASTERISK-29235</a>: res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address<br/>Reported by: Brian Paboojian<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=976b1a1d7ac4dfca60e43e9f36f73510b22b23d6">[976b1a1d7a]</a> Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on REGISTER responses.</li>
+</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29105">ASTERISK-29105</a>: chan_pjsip: 180 Ringing with SDP not changed into progress<br/>Reported by: Sebastian Damm<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3286c048568620810f283c706f443268c9920df6">[3286c04856]</a> Holger Hans Peter Freyther -- pjsip: Generate progress (once) when receiving a 180 with a SDP</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28452">ASTERISK-28452</a>: pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer<br/>Reported by: Michael Maier<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1af2a84c8bcc442374cccd09cd46c57854a209a2">[1af2a84c8b]</a> Joshua C. Colp -- res_pjsip_session: Always produce offer on re-INVITE without SDP.</li>
+</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29215">ASTERISK-29215</a>: res_pjsip_session: NULL active_media_state topology caused asterisk crash<br/>Reported by: sungtae kim<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bffff6e2d0312335ffa720f57de22ae7ca0a5de4">[bffff6e2d0]</a> George Joseph -- res_pjsip_session: Make reschedule_reinvite check for NULL topologies</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29303">ASTERISK-29303</a>: pjsip: Re-invite occurs when it shouldn't<br/>Reported by: Benjamin Keith Ford<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=83b0f5963ff3a3e598591afca58ea05b7c66441b">[83b0f5963f]</a> Ben Ford -- res_pjsip_session.c: Check topology on re-invite.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fad0cf12e6af0b5f193d825f8498159669473292">[fad0cf12e6]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29220">ASTERISK-29220</a>: After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used<br/>Reported by: Robert Cripps<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=017e09b40a8f09001750172fa13afc5e40c42d5c">[017e09b40a]</a> Robert Cripps -- res/res_pjsip_session.c: Check that media type matches in</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29248">ASTERISK-29248</a>: res_pjsip_session: res sometimes uninitialized reported by compiler Clang.<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f119192bbdb0c236b51831adfed7659a95fb194">[3f119192bb]</a> Alexander Traud -- res_pjsip_session: Avoid sometimes-uninitialized warning with Clang.</li>
+</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fad0cf12e6af0b5f193d825f8498159669473292">[fad0cf12e6]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
+</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29300">ASTERISK-29300</a>: res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent<br/>Reported by: Sebastian Damm<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90ef6a14a7d75421f6ead90cf6aa1d48b92543fc">[90ef6a14a7]</a> Torrey Searle -- res/res_rtp_asterisk: generate new SSRC on native bridge end</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29266">ASTERISK-29266</a>: ICE Role conflict with an unauthorized session<br/>Reported by: Salah Ahmed<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df8d335ad1bfd21f1154010ae4b60f596b80952a">[df8d335ad1]</a> Salah Ahmed -- res_rtp_asterisk:  Check remote ICE reset and reset local ice attrb</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29205">ASTERISK-29205</a>: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client<br/>Reported by: Edvin Vidmar<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a6f2f913b99613419f82b14aca60792760f7485">[5a6f2f913b]</a> Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch that leads to overflow</li>
+</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28452">ASTERISK-28452</a>: pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer<br/>Reported by: Michael Maier<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1af2a84c8bcc442374cccd09cd46c57854a209a2">[1af2a84c8b]</a> Joshua C. Colp -- res_pjsip_session: Always produce offer on re-INVITE without SDP.</li>
+</ul><br><h3>Improvement</h3><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29244">ASTERISK-29244</a>: Add MixMonitorStart / Stop / Mute AMI events<br/>Reported by: Sébastien Duthil<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=092628c9829f1ff89333f9c5099b649ec060b5ac">[092628c982]</a> Sebastien Duthil -- app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute.</li>
+</ul><br><h4>Category: Applications/app_transfer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29252">ASTERISK-29252</a>: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code<br/>Reported by: Dan Cropp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=088816284a51a72207344324671beb53ada56ae7">[088816284a]</a> Dan Cropp -- chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable</li>
+</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29252">ASTERISK-29252</a>: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code<br/>Reported by: Dan Cropp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=088816284a51a72207344324671beb53ada56ae7">[088816284a]</a> Dan Cropp -- chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable</li>
+</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29326">ASTERISK-29326</a>: asterisk: Update copyright/company<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=682f7d943752b0385f58ce4ad0bdcbf22babdbc0">[682f7d9437]</a> Joshua C. Colp -- asterisk: Update copyright.</li>
+</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29321">ASTERISK-29321</a>: sorcery: Add support for more intelligent reloading.<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9acbd19f37b1ef0c73cc929301ce39b23fd1df7">[a9acbd19f3]</a> Joshua C. Colp -- sorcery: Add support for more intelligent reloading.</li>
+</ul><br><h4>Category: Formats/format_wav</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29275">ASTERISK-29275</a>: Support of MIME-type for wav16<br/>Reported by: Boris P. Korzun<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=57d130d3aaab1a7ee06f5bd43ed7c191f3ad6b44">[57d130d3aa]</a> Boris P. Korzun -- format_wav: Support of MIME-type for wav16</li>
+</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29262">ASTERISK-29262</a>: Support of various URL-schemes by MoH<br/>Reported by: Boris P. Korzun<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1c88a497b83f57b1851cd598c1a19bea21af445">[f1c88a497b]</a> Boris P. Korzun -- res_musiconhold: Add support of various URL-schemes by MoH.</li>
+</ul><br><h4>Category: Resources/res_pjsip_registrar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29325">ASTERISK-29325</a>: res_pjsip_registrar: Include source IP address and port in log messages<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f1c21e4ca60393e52d2b242184402913c0a6632">[5f1c21e4ca]</a> Joshua C. Colp -- res_pjsip_registrar: Include source IP and port in log messages.</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=2c0e6bac06c3ce4b70b5425168db7a9417cac2eb">2c0e6bac06</a></td><td>Asterisk Development Team</td><td>Update for 18.3.0-rc2</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae4a3da5570e2c49a1f200700fde856de93f6e6c">ae4a3da557</a></td><td>Asterisk Development Team</td><td>Update for 18.3.0-rc1</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=263f906af468e987bcd37016ec52233dae8131be">263f906af4</a></td><td>Kevin Harwell</td><td>manager: Increase the non breaking AMI version number</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0afd37e3b500802dfd8b921d5c79f46fd2b0ef4d">0afd37e3b5</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.3.0</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=23e41313a846f2b7ff4ee6bebc91836474e366c2">23e41313a8</a></td><td>Jaco Kroon</td><td>func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52707fba7f41be762b0dd1614254434274312065">52707fba7f</a></td><td>Jaco Kroon</td><td>app.h: Fix -Werror=zero-length-bounds compile errors in dev mode.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=262473c6d9f2fee201cd48a2ae298eb0ce2b903e">262473c6d9</a></td><td>Alexander Traud</td><td>res_format_attr_*: Parameter Names are Case-Insensitive.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4fc0e16838df8ee2085f58bf33f8ff07316015b6">4fc0e16838</a></td><td>Alexander Traud</td><td>chan_iax2: System Header strings is included via asterisk.h/compat.h.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16e4d1f36fa6b0a567b0671d7225c1d9473a4c7e">16e4d1f36f</a></td><td>Sean Bright</td><td>res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=269bb08ea290cbdb677892ed6b20885112ee1c62">269bb08ea2</a></td><td>George Joseph</td><td>res_pjsip_refer: Move the progress dlg release to a serializer</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=032329314269ef753c1ba00d93cbc6370d003908">0323293142</a></td><td>Alexander Traud</td><td>res_format_attr_h263: Generate valid SDP fmtp for H.263+.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be0a61bc3d1709b45e39540f48c115898eca0b93">be0a61bc3d</a></td><td>Kevin Harwell</td><td>res_rtp_asterisk: Add packet subtype during RTCP debug when relevant</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1adf9368eeefd2889c95dcc122ab1281e33a0de4">1adf9368ee</a></td><td>Alexander Traud</td><td>chan_sip: Filter pass-through audio/video formats away, again.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bee35fe04ac9473f5d98588421fd1a2dad401cb8">bee35fe04a</a></td><td>Jaco Kroon</td><td>func_odbc:  Introduce minargs config and expose ARGC in addition to ARGn.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbd8908f8d13e6f6c62738ccb9ff6aee0f2d0c30">dbd8908f8d</a></td><td>George Joseph</td><td>res_pjsip_refer: Always serialize calls to refer_progress_notify</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28f187d6c5ae57a02569c86bf86b49ea5029d862">28f187d6c5</a></td><td>George Joseph</td><td>chan_iax2.c: Require secret and auth method if encryption is enabled</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=24d6adfe99bf4a14a384bb6541cd9927360cdaa0">24d6adfe99</a></td><td>Sean Bright</td><td>app_read: Release tone zone reference on early return.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c0fbaf010122159c8b2aeff213e7e16e4d4f7ee">7c0fbaf010</a></td><td>Ivan Poddubnyi</td><td>main/frame: Add missing control frame names to ast_frame_subclass2str</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb42b603261aa73cf68ea84e9f9478f588604547">fb42b60326</a></td><td>Sean Bright</td><td>res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c5687092953604821f16c417815b9254f340f87">9c56870929</a></td><td>Jaco Kroon</td><td>AC_HEADER_STDC causes a compile failure with autoconf 2.70</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a25bcf70ed9556c915ee96b11cfbad42f1462b1f">a25bcf70ed</a></td><td>Alexander Traud</td><td>pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87a35f8e945b5ba3beecfaac8c7cd2a3ff912e6c">87a35f8e94</a></td><td>Ben Ford</td><td>chan_pjsip.c: Add parameters to frame in indicate.</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-18.2.0-summary.html             |  169 -----
+asterisk-18.2.0-summary.txt              |  508 -----------------
+b/.version                               |    2
+b/CHANGES                                |   53 +
+b/ChangeLog                              |  923 ++++++++++++++++++++++++++++++-
+b/README.md                              |    8
+b/UPGRADE.txt                            |   14
+b/apps/app_dial.c                        |   14
+b/apps/app_mixmonitor.c                  |   75 ++
+b/apps/app_page.c                        |   13
+b/apps/app_queue.c                       |    6
+b/apps/app_read.c                        |    3
+b/apps/app_transfer.c                    |   24
+b/asterisk-18.3.0-rc2-summary.html       |   14
+b/asterisk-18.3.0-rc2-summary.txt        |  102 +++
+b/channels/chan_iax2.c                   |   40 +
+b/channels/chan_pjsip.c                  |   41 +
+b/channels/chan_sip.c                    |   60 --
+b/configs/samples/func_odbc.conf.sample  |   11
+b/configs/samples/iax.conf.sample        |    9
+b/configs/samples/modules.conf.sample    |   16
+b/configs/samples/rtp.conf.sample        |   12
+b/configs/samples/stasis.conf.sample     |    2
+b/configure                              |  116 ---
+b/configure.ac                           |    5
+b/formats/format_wav.c                   |    3
+b/funcs/func_callerid.c                  |  146 ++--
+b/funcs/func_odbc.c                      |   31 -
+b/include/asterisk/app.h                 |    7
+b/include/asterisk/channel.h             |   12
+b/include/asterisk/core_unreal.h         |    2
+b/include/asterisk/manager.h             |    2
+b/include/asterisk/sorcery.h             |   22
+b/include/asterisk/stasis_channels.h     |   26
+b/include/asterisk/strings.h             |    4
+b/main/asterisk.c                        |    8
+b/main/channel.c                         |   45 +
+b/main/core_unreal.c                     |   92 +++
+b/main/frame.c                           |    9
+b/main/manager_channels.c                |   56 +
+b/main/say.c                             |    4
+b/main/sorcery.c                         |   17
+b/main/stasis.c                          |    3
+b/main/stasis_channels.c                 |    9
+b/main/translate.c                       |   24
+b/res/res_agi.c                          |    6
+b/res/res_config_pgsql.c                 |   32 -
+b/res/res_fax.c                          |   12
+b/res/res_format_attr_celt.c             |   14
+b/res/res_format_attr_h263.c             |  141 ++++
+b/res/res_format_attr_ilbc.c             |   12
+b/res/res_format_attr_opus.c             |   31 -
+b/res/res_format_attr_silk.c             |   17
+b/res/res_format_attr_siren14.c          |   13
+b/res/res_format_attr_siren7.c           |   13
+b/res/res_format_attr_vp8.c              |   12
+b/res/res_musiconhold.c                  |   10
+b/res/res_odbc_transaction.c             |    5
+b/res/res_pjsip.c                        |    2
+b/res/res_pjsip/pjsip_scheduler.c        |    2
+b/res/res_pjsip_diversion.c              |   14
+b/res/res_pjsip_endpoint_identifier_ip.c |    3
+b/res/res_pjsip_nat.c                    |   24
+b/res/res_pjsip_outbound_registration.c  |   13
+b/res/res_pjsip_path.c                   |   12
+b/res/res_pjsip_pubsub.c                 |    2
+b/res/res_pjsip_refer.c                  |  163 +++--
+b/res/res_pjsip_registrar.c              |   21
+b/res/res_pjsip_sdp_rtp.c                |   42 +
+b/res/res_pjsip_session.c                |  197 +++---
+b/res/res_pjsip_t38.c                    |    9
+b/res/res_rtp_asterisk.c                 |   75 ++
+b/res/res_sorcery_config.c               |    6
+73 files changed, 2450 insertions(+), 1215 deletions(-)</pre><br></html>
--- a/asterisk-18.3.0-summary.txt
+++ b/asterisk-18.3.0-summary.txt
@@ -0,0 +1,592 @@
+                                Release Summary
+
+                                asterisk-18.3.0
+
+                                Date: 2021-03-25
+
+                           <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-18.2.0.
+
+     ----------------------------------------------------------------------
+
+                                  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                     
+   11 Alexander Traud           1 Mark Petersen 6 Alexander Traud             
+   9 Joshua C. Colp                             3 Boris P. Korzun             
+   8 Sean Bright                                3 Joshua C. Colp              
+   6 Jaco Kroon                                 2 Matthias Hensler            
+   5 George Joseph                              2 Stefan Ruf                  
+   4 Ben Ford                                   2 Sebastian Damm              
+   3 Kevin Harwell                              2 Gregory Massel              
+   3 Asterisk Development Team                  1 Rusty Newton                
+   3 Ivan Poddubnyi                             1 Alexei Gradinari            
+   3 Boris P. Korzun                            1 Ivan Poddubny               
+   1 Salah Ahmed                                1 Jacek Konieczny             
+   1 Dan Cropp                                  1 Jaco Kroon                  
+   1 Holger Hans Peter Freyther                 1 Edvin Vidmar                
+   1 Nico Kooijman                              1 Sébastien Duthil            
+   1 Alexei Gradinari                           1 Jean Aunis - Prescom        
+   1 Torrey Searle                              1 sungtae kim                 
+   1 Nick French                                1 Benjamin Keith Ford         
+   1 Robert Cripps                              1 Boolah                      
+   1 Sebastien Duthil                           1 Nick French                 
+   1 Mark Petersen                              1 Salah Ahmed                 
+                                                1 Mauri de Souza Meneguzzo    
+                                                (3CPlus)                      
+                                                1 N A                         
+                                                1 N A                         
+                                                1 Jacek Konieczny             
+                                                1 IAMJames_                   
+                                                1 Mark Petersen               
+                                                1 Dan Cropp                   
+                                                1 Ivan Poddubny               
+                                                1 Vitezslav Novy              
+                                                1 Mark Petersen               
+                                                1 Michael Maier               
+                                                1 George Joseph               
+                                                1 Alexander Traud             
+                                                1 Brian Paboojian             
+                                                1 Dan Cropp                   
+                                                1 Robert Cripps               
+
+     ----------------------------------------------------------------------
+
+                                 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: Resources/res_pjsip_t38
+
+   ASTERISK-29305: ASTERISK-29203 / AST-2021-002 -- Another scenario is
+   causing a crash
+   Reported by: Gregory Massel
+     * [77328142b4] Ben Ford -- AST-2021-006 - res_pjsip_t38.c: Check for
+       session_media on reinvite.
+
+    Category: Resources/res_srtp
+
+   ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
+   Reported by: Alexander Traud
+     * [703158b903] Alexander Traud -- rtp: Enable srtp replay protection
+
+    Category: pjproject/pjsip
+
+   ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes
+   memory corruption and crash
+   Reported by: Ivan Poddubny
+     * [2770cc5872] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more
+       than one histinfo to Supported
+
+  Bug
+
+    Category: Applications/General
+
+   ASTERISK-29287: app.h: C++ compatibility broken
+   Reported by: Jean Aunis - Prescom
+     * [916d5d5e45] Jaco Kroon -- app.h: Restore C++ compatibility for macro
+       AST_DECLARE_APP_ARGS
+
+    Category: Applications/app_confbridge
+
+   ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
+   muting over AMI occurs
+   Reported by: Stefan Ruf
+     * [f7bda066bb] Joshua C. Colp -- channel: Fix crash in suppress API.
+     * [b43b81d953] Joshua C. Colp -- channel: Fix memory leak in suppress
+       API.
+
+    Category: Applications/app_dial
+
+   ASTERISK-29329: app_dial: DTMF to 'D' option gets duplicated if there are
+   multiple progress events
+   Reported by: N A
+     * [94debe5085] Sean Bright -- app_dial.c: Only send DTMF on first
+       progress event.
+
+    Category: Applications/app_page
+
+   ASTERISK-16799: Callee declined when 'beep' audio file does not exist
+   Reported by: IAMJames_
+     * [6673c1b177] Sean Bright -- app_page.c: Don't fail to Page if beep
+       sound file is missing
+
+    Category: Applications/app_queue
+
+   ASTERISK-28369: app_queue: Member device state "invalid" when second call
+   is ringing and hint is used
+   Reported by: Boolah
+     * [985d3e4940] Ivan Poddubnyi -- app_queue: Fix conversion of complex
+       extension states into device states
+
+    Category: Channels/chan_local
+
+   ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing
+   Reported by: Matthias Hensler
+     * [47e9ce96ea] Joshua C. Colp -- core_unreal: Fix deadlock with T.38
+       control frames.
+     * [62e2dd484d] Ben Ford -- core_unreal: Fix T.38 faxing when using local
+       channels.
+
+    Category: Channels/chan_sip/CodecHandling
+
+   ASTERISK-29280: chan_sip: Allow peers without audio (text+video).
+   Reported by: Alexander Traud
+     * [45e48e387c] Alexander Traud -- chan_sip: Allow [peer] without audio
+       (text+video).
+   ASTERISK-29265: chan_sip: Allow text+video media streams, again.
+   Reported by: Alexander Traud
+     * [87ad1138ff] Alexander Traud -- chan_sip: Set up calls without audio
+       (text+video), again.
+   ASTERISK-29258: chan_sip: Audio stream rejected, Other stream present:
+   Invalid SDP.
+   Reported by: Alexander Traud
+     * [4c154f3431] Alexander Traud -- chan_sip: SDP: Reject audio streams
+       correctly.
+
+    Category: Core/Bridging
+
+   ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
+   muting over AMI occurs
+   Reported by: Stefan Ruf
+     * [f7bda066bb] Joshua C. Colp -- channel: Fix crash in suppress API.
+     * [b43b81d953] Joshua C. Colp -- channel: Fix memory leak in suppress
+       API.
+
+    Category: Core/Channels
+
+   ASTERISK-29259: channel: Allow text+video media streams, again.
+   Reported by: Alexander Traud
+     * [f64ddf3db3] Alexander Traud -- channel: Set up calls without audio
+       (text+video), again.
+
+    Category: Core/General
+
+   ASTERISK-29306: strings: Incorrect use of __attribute__((pure)) in
+   ast_str_to_lower definition
+   Reported by: Vitezslav Novy
+     * [e4cd7a7d0b] Sean Bright -- strings.h: ast_str_to_upper() and
+       _to_lower() are not pure.
+
+    Category: Core/Internationalization
+
+   ASTERISK-29297: say: Y2021 problem – Asterisk cannot say year 2021 in
+   Dutch
+   Reported by: Jacek Konieczny
+     * [7b052ec965] Nico Kooijman -- main: With Dutch language year after
+       2020 is not spoken in say.c
+
+    Category: Documentation
+
+   ASTERISK-24434: Fix differing usage of assignment operators in
+   modules.conf
+   Reported by: Rusty Newton
+     * [3084084648] Sean Bright -- modules.conf: Fix differing usage of
+       assignment operators.
+
+    Category: Resources/res_config_pgsql
+
+   ASTERISK-29293: res_config_pgsql: Limit realtime_pgsql() to return one (no
+   more) record
+   Reported by: Boris P. Korzun
+     * [beb579bc99] Boris P. Korzun -- res_config_pgsql: Limit
+       realtime_pgsql() to return one (no more) record.
+
+    Category: Resources/res_fax
+
+   ASTERISK-29312: res_fax: asterisk fails to publish the Stasis and
+   ReceiveFax status messages if the remote Station ID contains invalid UTF-8
+   characters
+   Reported by: Alexei Gradinari
+     * [d5e73d2121] Alexei Gradinari -- res_fax: validate the remote/local
+       Station ID for UTF-8 format
+
+    Category: Resources/res_odbc
+
+   ASTERISK-29311: res_odbc_transaction sets forcecommit default value based
+   on isolation level instead of forcecommit
+   Reported by: Jaco Kroon
+     * [7ab53fce7a] Jaco Kroon -- res_odbc_transaction: correctly initialise
+       forcecommit value from DSN.
+
+    Category: Resources/res_pjsip
+
+   ASTERISK-29196: res_pjsip: Segmentation fault
+   Reported by: Mauri de Souza Meneguzzo (3CPlus)
+     * [acb7ce4fe7] Joshua C. Colp -- pjsip: Make modify_local_offer2
+       tolerate previous failed SDP.
+   ASTERISK-29261: res_pjsip: user=phone validation fail for isup numbers
+   containing *#
+   Reported by: Mark Petersen
+     * [176274caa4] Mark Petersen -- res/res_pjsip.c: allow user=phone when
+       number contain *#
+
+    Category: Resources/res_pjsip_nat
+
+   ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
+   with external_signaling_address
+   Reported by: Brian Paboojian
+     * [976b1a1d7a] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
+       REGISTER responses.
+
+    Category: Resources/res_pjsip_outbound_registration
+
+   ASTERISK-29315: res_pjsip: re-registration gets stuck if setting initial
+   auth credentials fails
+   Reported by: Nick French
+     * [dedfb334bd] Nick French -- res_pjsip: dont return early from
+       registration if init auth fails
+
+    Category: Resources/res_pjsip_refer
+
+   ASTERISK-29313: res_pjsip_refer: Segfault in progress notify
+   Reported by: George Joseph
+     * [15afabdf8e] George Joseph -- res_pjsip_refer: Refactor progress
+       locking and serialization
+
+    Category: Resources/res_pjsip_registrar
+
+   ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
+   with external_signaling_address
+   Reported by: Brian Paboojian
+     * [976b1a1d7a] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
+       REGISTER responses.
+
+    Category: Resources/res_pjsip_sdp_rtp
+
+   ASTERISK-29105: chan_pjsip: 180 Ringing with SDP not changed into progress
+   Reported by: Sebastian Damm
+     * [3286c04856] Holger Hans Peter Freyther -- pjsip: Generate progress
+       (once) when receiving a 180 with a SDP
+   ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
+   on reinvite without SDP offer
+   Reported by: Michael Maier
+     * [1af2a84c8b] Joshua C. Colp -- res_pjsip_session: Always produce offer
+       on re-INVITE without SDP.
+
+    Category: Resources/res_pjsip_session
+
+   ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
+   asterisk crash
+   Reported by: sungtae kim
+     * [bffff6e2d0] George Joseph -- res_pjsip_session: Make
+       reschedule_reinvite check for NULL topologies
+   ASTERISK-29303: pjsip: Re-invite occurs when it shouldn't
+   Reported by: Benjamin Keith Ford
+     * [83b0f5963f] Ben Ford -- res_pjsip_session.c: Check topology on
+       re-invite.
+   ASTERISK-29203: res_pjsip_t38: Crash when changing state
+   Reported by: Gregory Massel
+     * [fad0cf12e6] Kevin Harwell -- AST-2021-002: Remote crash possible when
+       negotiating T.38
+   ASTERISK-29220: After T38 reinvite response of 488 a subsequent G711
+   reinvite is not processed correctly. Instead the previous T38 session
+   media is used
+   Reported by: Robert Cripps
+     * [017e09b40a] Robert Cripps -- res/res_pjsip_session.c: Check that
+       media type matches in
+   ASTERISK-29248: res_pjsip_session: res sometimes uninitialized reported by
+   compiler Clang.
+   Reported by: Alexander Traud
+     * [3f119192bb] Alexander Traud -- res_pjsip_session: Avoid
+       sometimes-uninitialized warning with Clang.
+
+    Category: Resources/res_pjsip_t38
+
+   ASTERISK-29203: res_pjsip_t38: Crash when changing state
+   Reported by: Gregory Massel
+     * [fad0cf12e6] Kevin Harwell -- AST-2021-002: Remote crash possible when
+       negotiating T.38
+
+    Category: Resources/res_rtp_asterisk
+
+   ASTERISK-29300: res_rtp_asterisk: When native local bridging the remote
+   SSRC becomes permanent
+   Reported by: Sebastian Damm
+     * [90ef6a14a7] Torrey Searle -- res/res_rtp_asterisk: generate new SSRC
+       on native bridge end
+   ASTERISK-29266: ICE Role conflict with an unauthorized session
+   Reported by: Salah Ahmed
+     * [df8d335ad1] Salah Ahmed -- res_rtp_asterisk: Check remote ICE reset
+       and reset local ice attrb
+   ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold
+   from webrtc client
+   Reported by: Edvin Vidmar
+     * [5a6f2f913b] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch
+       that leads to overflow
+
+    Category: pjproject/pjsip
+
+   ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
+   on reinvite without SDP offer
+   Reported by: Michael Maier
+     * [1af2a84c8b] Joshua C. Colp -- res_pjsip_session: Always produce offer
+       on re-INVITE without SDP.
+
+  Improvement
+
+    Category: Applications/app_mixmonitor
+
+   ASTERISK-29244: Add MixMonitorStart / Stop / Mute AMI events
+   Reported by: Sébastien Duthil
+     * [092628c982] Sebastien Duthil -- app_mixmonitor: Add AMI events
+       MixMonitorStart, -Stop and -Mute.
+
+    Category: Applications/app_transfer
+
+   ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
+   failure SIP code
+   Reported by: Dan Cropp
+     * [088816284a] Dan Cropp -- chan_pjsip, app_transfer: Add
+       TRANSFERSTATUSPROTOCOL variable
+
+    Category: Channels/chan_pjsip
+
+   ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
+   failure SIP code
+   Reported by: Dan Cropp
+     * [088816284a] Dan Cropp -- chan_pjsip, app_transfer: Add
+       TRANSFERSTATUSPROTOCOL variable
+
+    Category: Core/General
+
+   ASTERISK-29326: asterisk: Update copyright/company
+   Reported by: Joshua C. Colp
+     * [682f7d9437] Joshua C. Colp -- asterisk: Update copyright.
+
+    Category: Core/Sorcery
+
+   ASTERISK-29321: sorcery: Add support for more intelligent reloading.
+   Reported by: Joshua C. Colp
+     * [a9acbd19f3] Joshua C. Colp -- sorcery: Add support for more
+       intelligent reloading.
+
+    Category: Formats/format_wav
+
+   ASTERISK-29275: Support of MIME-type for wav16
+   Reported by: Boris P. Korzun
+     * [57d130d3aa] Boris P. Korzun -- format_wav: Support of MIME-type for
+       wav16
+
+    Category: Resources/res_musiconhold
+
+   ASTERISK-29262: Support of various URL-schemes by MoH
+   Reported by: Boris P. Korzun
+     * [f1c88a497b] Boris P. Korzun -- res_musiconhold: Add support of
+       various URL-schemes by MoH.
+
+    Category: Resources/res_pjsip_registrar
+
+   ASTERISK-29325: res_pjsip_registrar: Include source IP address and port in
+   log messages
+   Reported by: Joshua C. Colp
+     * [5f1c21e4ca] Joshua C. Colp -- res_pjsip_registrar: Include source IP
+       and port in log messages.
+
+     ----------------------------------------------------------------------
+
+                      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                                |
+   |------------+------------------+----------------------------------------|
+   | 2c0e6bac06 | Asterisk         | Update for 18.3.0-rc2                  |
+   |            | Development Team |                                        |
+   |------------+------------------+----------------------------------------|
+   | ae4a3da557 | Asterisk         | Update for 18.3.0-rc1                  |
+   |            | Development Team |                                        |
+   |------------+------------------+----------------------------------------|
+   | 263f906af4 | Kevin Harwell    | manager: Increase the non breaking AMI |
+   |            |                  | version number                         |
+   |------------+------------------+----------------------------------------|
+   | 0afd37e3b5 | Asterisk         | Update CHANGES and UPGRADE.txt for     |
+   |            | Development Team | 18.3.0                                 |
+   |------------+------------------+----------------------------------------|
+   |            |                  | func_callerid+res_agi: Fix compile     |
+   | 23e41313a8 | Jaco Kroon       | errors related to                      |
+   |            |                  | -Werror=zero-length-bounds             |
+   |------------+------------------+----------------------------------------|
+   | 52707fba7f | Jaco Kroon       | app.h: Fix -Werror=zero-length-bounds  |
+   |            |                  | compile errors in dev mode.            |
+   |------------+------------------+----------------------------------------|
+   | 262473c6d9 | Alexander Traud  | res_format_attr_*: Parameter Names are |
+   |            |                  | Case-Insensitive.                      |
+   |------------+------------------+----------------------------------------|
+   | 4fc0e16838 | Alexander Traud  | chan_iax2: System Header strings is    |
+   |            |                  | included via asterisk.h/compat.h.      |
+   |------------+------------------+----------------------------------------|
+   | 16e4d1f36f | Sean Bright      | res_musiconhold.c: Plug ref leak       |
+   |            |                  | caused by ao2_replace() misuse.        |
+   |------------+------------------+----------------------------------------|
+   | 269bb08ea2 | George Joseph    | res_pjsip_refer: Move the progress dlg |
+   |            |                  | release to a serializer                |
+   |------------+------------------+----------------------------------------|
+   | 0323293142 | Alexander Traud  | res_format_attr_h263: Generate valid   |
+   |            |                  | SDP fmtp for H.263+.                   |
+   |------------+------------------+----------------------------------------|
+   | be0a61bc3d | Kevin Harwell    | res_rtp_asterisk: Add packet subtype   |
+   |            |                  | during RTCP debug when relevant        |
+   |------------+------------------+----------------------------------------|
+   | 1adf9368ee | Alexander Traud  | chan_sip: Filter pass-through          |
+   |            |                  | audio/video formats away, again.       |
+   |------------+------------------+----------------------------------------|
+   | bee35fe04a | Jaco Kroon       | func_odbc: Introduce minargs config    |
+   |            |                  | and expose ARGC in addition to ARGn.   |
+   |------------+------------------+----------------------------------------|
+   | dbd8908f8d | George Joseph    | res_pjsip_refer: Always serialize      |
+   |            |                  | calls to refer_progress_notify         |
+   |------------+------------------+----------------------------------------|
+   | 28f187d6c5 | George Joseph    | chan_iax2.c: Require secret and auth   |
+   |            |                  | method if encryption is enabled        |
+   |------------+------------------+----------------------------------------|
+   | 24d6adfe99 | Sean Bright      | app_read: Release tone zone reference  |
+   |            |                  | on early return.                       |
+   |------------+------------------+----------------------------------------|
+   | 7c0fbaf010 | Ivan Poddubnyi   | main/frame: Add missing control frame  |
+   |            |                  | names to ast_frame_subclass2str        |
+   |------------+------------------+----------------------------------------|
+   | fb42b60326 | Sean Bright      | res_pjsip_pubsub: Fix truncation of    |
+   |            |                  | persisted SUBSCRIBE packet             |
+   |------------+------------------+----------------------------------------|
+   | 9c56870929 | Jaco Kroon       | AC_HEADER_STDC causes a compile        |
+   |            |                  | failure with autoconf 2.70             |
+   |------------+------------------+----------------------------------------|
+   |            |                  | pjsip_scheduler: Fix pjsip show        |
+   | a25bcf70ed | Alexander Traud  | scheduled_tasks like for compiler      |
+   |            |                  | Clang.                                 |
+   |------------+------------------+----------------------------------------|
+   | 87a35f8e94 | Ben Ford         | chan_pjsip.c: Add parameters to frame  |
+   |            |                  | in indicate.                           |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                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-18.2.0-summary.html             |  169 -----
+ asterisk-18.2.0-summary.txt              |  508 -----------------
+ b/.version                               |    2
+ b/CHANGES                                |   53 +
+ b/ChangeLog                              |  923 ++++++++++++++++++++++++++++++-
+ b/README.md                              |    8
+ b/UPGRADE.txt                            |   14
+ b/apps/app_dial.c                        |   14
+ b/apps/app_mixmonitor.c                  |   75 ++
+ b/apps/app_page.c                        |   13
+ b/apps/app_queue.c                       |    6
+ b/apps/app_read.c                        |    3
+ b/apps/app_transfer.c                    |   24
+ b/asterisk-18.3.0-rc2-summary.html       |   14
+ b/asterisk-18.3.0-rc2-summary.txt        |  102 +++
+ b/channels/chan_iax2.c                   |   40 +
+ b/channels/chan_pjsip.c                  |   41 +
+ b/channels/chan_sip.c                    |   60 --
+ b/configs/samples/func_odbc.conf.sample  |   11
+ b/configs/samples/iax.conf.sample        |    9
+ b/configs/samples/modules.conf.sample    |   16
+ b/configs/samples/rtp.conf.sample        |   12
+ b/configs/samples/stasis.conf.sample     |    2
+ b/configure                              |  116 ---
+ b/configure.ac                           |    5
+ b/formats/format_wav.c                   |    3
+ b/funcs/func_callerid.c                  |  146 ++--
+ b/funcs/func_odbc.c                      |   31 -
+ b/include/asterisk/app.h                 |    7
+ b/include/asterisk/channel.h             |   12
+ b/include/asterisk/core_unreal.h         |    2
+ b/include/asterisk/manager.h             |    2
+ b/include/asterisk/sorcery.h             |   22
+ b/include/asterisk/stasis_channels.h     |   26
+ b/include/asterisk/strings.h             |    4
+ b/main/asterisk.c                        |    8
+ b/main/channel.c                         |   45 +
+ b/main/core_unreal.c                     |   92 +++
+ b/main/frame.c                           |    9
+ b/main/manager_channels.c                |   56 +
+ b/main/say.c                             |    4
+ b/main/sorcery.c                         |   17
+ b/main/stasis.c                          |    3
+ b/main/stasis_channels.c                 |    9
+ b/main/translate.c                       |   24
+ b/res/res_agi.c                          |    6
+ b/res/res_config_pgsql.c                 |   32 -
+ b/res/res_fax.c                          |   12
+ b/res/res_format_attr_celt.c             |   14
+ b/res/res_format_attr_h263.c             |  141 ++++
+ b/res/res_format_attr_ilbc.c             |   12
+ b/res/res_format_attr_opus.c             |   31 -
+ b/res/res_format_attr_silk.c             |   17
+ b/res/res_format_attr_siren14.c          |   13
+ b/res/res_format_attr_siren7.c           |   13
+ b/res/res_format_attr_vp8.c              |   12
+ b/res/res_musiconhold.c                  |   10
+ b/res/res_odbc_transaction.c             |    5
+ b/res/res_pjsip.c                        |    2
+ b/res/res_pjsip/pjsip_scheduler.c        |    2
+ b/res/res_pjsip_diversion.c              |   14
+ b/res/res_pjsip_endpoint_identifier_ip.c |    3
+ b/res/res_pjsip_nat.c                    |   24
+ b/res/res_pjsip_outbound_registration.c  |   13
+ b/res/res_pjsip_path.c                   |   12
+ b/res/res_pjsip_pubsub.c                 |    2
+ b/res/res_pjsip_refer.c                  |  163 +++--
+ b/res/res_pjsip_registrar.c              |   21
+ b/res/res_pjsip_sdp_rtp.c                |   42 +
+ b/res/res_pjsip_session.c                |  197 +++---
+ b/res/res_pjsip_t38.c                    |    9
+ b/res/res_rtp_asterisk.c                 |   75 ++
+ b/res/res_sorcery_config.c               |    6
+ 73 files changed, 2450 insertions(+), 1215 deletions(-)
--- a/contrib/realtime/mysql/mysql_cdr.sql
+++ b/contrib/realtime/mysql/mysql_cdr.sql
@@ -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';
+
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
--- a/contrib/realtime/mysql/mysql_voicemail.sql
+++ b/contrib/realtime/mysql/mysql_voicemail.sql
@@ -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';
+
--- a/contrib/realtime/postgresql/postgresql_cdr.sql
+++ b/contrib/realtime/postgresql/postgresql_cdr.sql
@@ -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;
+
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
--- a/contrib/realtime/postgresql/postgresql_voicemail.sql
+++ b/contrib/realtime/postgresql/postgresql_voicemail.sql
@@ -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;
+
--- a/main/core_unreal.c
+++ b/main/core_unreal.c
@@ -710,6 +710,7 @@ int ast_unreal_indicate(struct ast_channel *ast, int condition, const void *data
 			 * signaling and we need to be sure that the locking order is the same to prevent possible
 			 * deadlocks.
 			 */
+			ast_channel_unlock(ast);
 			ast_unreal_lock_all(p, &chan, &owner);

 			if (owner) {
@@ -726,12 +727,14 @@ int ast_unreal_indicate(struct ast_channel *ast, int condition, const void *data
 			}

 			ao2_unlock(p);
+			ast_channel_lock(ast);
 		} else if (parameters->request_response == AST_T38_TERMINATED) {
 			/*
 			 * Lock both parts of the local channel so we can restore their topologies to the original.
 			 * The topology should be on the unreal_pvt structure, with a ref that we can steal. Same
 			 * conditions as above.
 			 */
+			ast_channel_unlock(ast);
 			ast_unreal_lock_all(p, &chan, &owner);

 			if (owner) {
@@ -745,6 +748,7 @@ int ast_unreal_indicate(struct ast_channel *ast, int condition, const void *data
 			}

 			ao2_unlock(p);
+			ast_channel_lock(ast);
 		}

 		/*
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -4273,18 +4273,40 @@ static void reschedule_reinvite(struct ast_sip_session *session, ast_sip_session
 	struct ast_sip_session_media_state *pending_media_state = NULL;
 	struct ast_sip_session_media_state *active_media_state = NULL;
 	const char *session_name = ast_sip_session_get_name(session);
+	int use_pending = 0;
+	int use_active = 0;
+
 	SCOPE_ENTER(3, "%s\n", session_name);

-	/* If the two media state topologies are the same this means that the session refresh request
+	/*
+	 * If the two media state topologies are the same this means that the session refresh request
 	 * did not specify a desired topology, so it does not care. If that is the case we don't even
-	 * pass one in here resulting in the current topology being used.
+	 * pass one in here resulting in the current topology being used.  It's possible though that
+	 * either one of the topologies could be NULL so we have to test for that before we check for
+	 * equality.
 	 */
-	if (!ast_stream_topology_equal(session->active_media_state->topology, session->pending_media_state->topology)) {
+
+	/* We only want to clone a media state if its topology is not null */
+	use_pending = session->pending_media_state->topology != NULL;
+	use_active = session->active_media_state->topology != NULL;
+
+	/*
+	 * If both media states have topologies, we can test for equality.  If they're equal we're not going to
+	 * clone either states.
+	 */
+	if (use_pending && use_active && ast_stream_topology_equal(session->active_media_state->topology, session->pending_media_state->topology)) {
+		use_pending = 0;
+		use_active = 0;
+	}
+
+	if (use_pending) {
 		pending_media_state = ast_sip_session_media_state_clone(session->pending_media_state);
 		if (!pending_media_state) {
 			SCOPE_EXIT_LOG_RTN(LOG_ERROR, "%s: Failed to clone pending media state\n", session_name);
 		}
+	}

+	if (use_active) {
 		active_media_state = ast_sip_session_media_state_clone(session->active_media_state);
 		if (!active_media_state) {
 			ast_sip_session_media_state_free(pending_media_state);
Author	SHA1	Message	Date
Asterisk Development Team	cd5f686bc3	Update for 18.3.0	2021-03-25 12:34:21 -05:00
Asterisk Development Team	2c0e6bac06	Update for 18.3.0-rc2	2021-03-22 10:41:13 -05:00
George Joseph	bffff6e2d0	res_pjsip_session: Make reschedule_reinvite check for NULL topologies When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	2021-03-22 09:40:32 -05:00
Joshua C. Colp	47e9ce96ea	core_unreal: Fix deadlock with T.38 control frames. When using the ast_unreal_lock_all function no channel locks can be held before calling it. This change unlocks the channel that indicate was called on before doing so and then relocks it afterwards. ASTERISK-29035 Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9	2021-03-19 09:02:19 -06:00
Asterisk Development Team	ae4a3da557	Update for 18.3.0-rc1	2021-03-11 12:23:06 -05:00