Update for 16.2.0-rc1

2025-10-14 08:31:02 +00:00 · 2019-02-06 06:08:57 -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-16.2.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.2.0-rc1</h3><h3 align="center">Date: 2019-02-06</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="#open_issues">Open Issues</a></li>
+<li><a href="#commits">Other Changes</a></li>
+<li><a href="#diffstat">Diffstat</a></li>
+</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.1.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">10 George Joseph <gjoseph@digium.com><br/>10 Sean Bright <sean.bright@gmail.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Alexei Gradinari <alex2grad@gmail.com><br/>2 Joshua C. Colp <jcolp@digium.com><br/>2 Jeremy Lainé <jeremy.laine@m4x.org><br/>2 Giuseppe Sucameli <sucameli@netresults.it><br/>2 Joshua Colp <jcolp@digium.com><br/>2 Chris-Savinovich <csavinovich@digium.com><br/>2 Richard Mudgett <rmudgett@digium.com><br/>1 Xiemin Chen <chenxiemin@gmail.com><br/>1 Mohit Dhiman <mohitdhiman@drishti-soft.com><br/>1 Pirmin Walthert <infos@nappsoft.ch><br/>1 Sungtae Kim <pchero21@gmail.com><br/>1 Diederik de Groot <dkgroot@talon.nl><br/>1 David M. Lee <dlee@respoke.io><br/>1 Jean Aunis <jean.aunis@prescom.fr><br/>1 Corey Farrell <git@cfware.com><br/>1 Bryan Boatright <ast-bugs@omega71.com><br/>1 Valentin Vidic <vvidic@valentin-vidic.from.hr><br/>1 sungtae kim <sungtae@messagebird.com><br/>1 Gerald Schnabel <gs@starface.de><br/>1 Ben Ford <bford@digium.com><br/>1 eyalhasson <eyal@kolhl.com><br/>1 Sebastian Damm <damm@sipgate.de><br/></td><td width="33%"><td width="33%">4 Joshua C. Colp <jcolp@digium.com><br/>3 George Joseph <gjoseph@digium.com><br/>2 Alexei Gradinari <alex2grad@gmail.com><br/>2 Giuseppe Sucameli <sucameli@netresults.it><br/>2 Jeremy Lainé <jeremy.laine@m4x.org><br/>2 David Kuehling <dvdkhlng@posteo.de><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Andrew Nagy<br/>1 boatright <ast-bugs@omega71.com><br/>1 Mohit Dhiman <mohitdhiman@drishti-soft.com><br/>1 sungtae kim <pchero21@gmail.com><br/>1 Ray <rainolf@gmail.com><br/>1 Eyal Hasson <eyal@kolhl.com><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 abelbeck <lonnie@abelbeck.com><br/>1 nappsoft <infos@nappsoft.ch><br/>1 Gianluca Merlo <gianluca.merlo@gmail.com><br/>1 Xiemin Chen <chenxiemin@gmail.com><br/>1 David Wilcox <david.wilcox@cloverbeen.com><br/>1 Andrew Nagy <andrew.nagy@the159.com><br/>1 Mark <wiewel@woop.la><br/>1 Diederik de Groot <dkgroot@talon.nl><br/>1 Valentin Vidić <vvidic@valentin-vidic.from.hr><br/>1 Gerald Schnabel <gs@starface.de><br/>1 xiemchen<br/>1 David Wilcox<br/>1 Sebastian Damm <damm@sipgate.de><br/>1 David Kuehling<br/></td></tr>
+</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: . I did not set the category correctly.</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28221">ASTERISK-28221</a>: Bug in ast_coredumper<br/>Reported by: Andrew Nagy<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3efe5061d5d0aac4c52843e6a3804e9212b10677">[3efe5061d5]</a> George Joseph -- ast_coredumper:  Refactor the pid determination process</li>
+</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28201">ASTERISK-28201</a>: [patch] confbridge: no announce to the marked users when they join an empty conference<br/>Reported by: Alexei Gradinari<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2610379605a48fd43afb1c9d89d9d797a81011df">[2610379605]</a> Alexei Gradinari -- confbridge: announce to the marked users when they join an empty conference</li>
+</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28218">ASTERISK-28218</a>: app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b)<br/>Reported by: Mark<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d9482695d20e4f7d6d5835dbfc1c11d728fe852">[2d9482695d]</a> Joshua Colp -- app_queue: Fix crash when using 'b' option on non-ringall queue.</li>
+</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28225">ASTERISK-28225</a>: app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"<br/>Reported by: boatright<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92298434bd9ce591356cda25b556c930a97a75f4">[92298434bd]</a> Bryan Boatright -- app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28222">ASTERISK-28222</a>: Regression: MWI polling no longer works<br/>Reported by: abelbeck<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff2ed4eeeee4837820ac23d061f03db5a61f5ec6">[ff2ed4eeee]</a> George Joseph -- Revert "stasis_cache:  Stop caching stasis subscription change messages"</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28215">ASTERISK-28215</a>: app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aebb822d1f5604a376f78e1d8ae555580aad4d64">[aebb822d1f]</a> George Joseph -- app_voicemail:  Don't delete mailbox state unless mailbox is deleted</li>
+</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28238">ASTERISK-28238</a>: PJSIP realtime. getcontext not working with DUNDI<br/>Reported by: Ray<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c3b4dcf807c06447ca4d38c6959b1a2561f60ff">[9c3b4dcf80]</a> Kevin Harwell -- pjsip/config_global: regcontext context not created</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28213">ASTERISK-28213</a>: res_pjsip: Threads pile up needlessly when AOR is blocked<br/>Reported by: Ross Beer<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1fb249132b1554e09e45a8a5f569ecc8b752568">[f1fb249132]</a> Kevin Harwell -- res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27095">ASTERISK-27095</a>: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5de36abd5af0b94a9fab1b8b51c5d1d90a95697a">[5de36abd5a]</a> Pirmin Walthert -- pjproject_bundled: check whether UPDATE is supported on outgoing calls</li>
+</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28194">ASTERISK-28194</a>: chan_sip: Leak using contact ACL<br/>Reported by: Giuseppe Sucameli<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6071ad77f5705a8f4cb3b41847955d92cd265a09">[6071ad77f5]</a> Giuseppe Sucameli -- chan_sip: Fix leak using contact ACL</li>
+</ul><br><h4>Category: Channels/chan_sip/Subscriptions</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28173">ASTERISK-28173</a>: Deadlock in chan_sip handling subscribe request during res_parking reload<br/>Reported by: Giuseppe Sucameli<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=419db481d1b44091bef52f17cc7acd7957d96c22">[419db481d1]</a> Giuseppe Sucameli -- Fix deadlock handling subscribe req during res_parking reload</li>
+</ul><br><h4>Category: Codecs/codec_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28263">ASTERISK-28263</a>: codec_opus: errors setting max_playback_rate and bitrate to "sdp"<br/>Reported by: Gianluca Merlo<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6452f9656cb42b588355fcb698ff23db5dba0b7">[f6452f9656]</a> Kevin Harwell -- codecs.conf.sample: update codec opus docs</li>
+</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28250">ASTERISK-28250</a>: build: Cross-compilation fails for target arm-linux-gnueabihf<br/>Reported by: Jean Aunis - Prescom<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3a6714158103120aeeffd62df662799858b0654">[d3a6714158]</a> Jean Aunis -- build : Fix cross-compilation errors</li>
+</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28197">ASTERISK-28197</a>: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases<br/>Reported by: Mohit Dhiman<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b24da607e695897f54f4b21208885fae7ac9158">[4b24da607e]</a> Mohit Dhiman -- stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.</li>
+</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28232">ASTERISK-28232</a>: core: RAII using clang use-after-scope issue<br/>Reported by: Diederik de Groot<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2c182b6ab63b8c39597f657ff7168c7d3424c8c">[d2c182b6ab]</a> Diederik de Groot -- RAII: Change order or variables in clang version</li>
+</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28252">ASTERISK-28252</a>: HangupHandler manager events are never thrown<br/>Reported by: Gerald Schnabel<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=735bd4d18576f43134dd17502d9c037bea996e81">[735bd4d185]</a> Gerald Schnabel -- manager_channels: Fix throwing of HangupHandler manager events</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28244">ASTERISK-28244</a>: stasis: Filter messages at publishing to AMI/ARI<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fcd07c34fbaa764cee47204a6bea838b8f4a3a27">[fcd07c34fb]</a> Joshua C. Colp -- stasis / manager / ari: Better filter messages.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28197">ASTERISK-28197</a>: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases<br/>Reported by: Mohit Dhiman<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b24da607e695897f54f4b21208885fae7ac9158">[4b24da607e]</a> Mohit Dhiman -- stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28212">ASTERISK-28212</a>: stasis: Statistics broke ABI under developer mode<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44a7faca21bf29bd7e09404d5dd200a0c8e95a8f">[44a7faca21]</a> Corey Farrell -- stasis: Fix ABI between DEVMODE and non-DEVMODE.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28117">ASTERISK-28117</a>: stasis: Add statistics for usage when in developer mode<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68ec7d93e82e02a37bacf2e2cc5c4ac0ea4d23c1">[68ec7d93e8]</a> Joshua C. Colp -- stasis: Add statistics gathering in developer mode.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28186">ASTERISK-28186</a>: stasis: Filter messages at publishing based on to_* presence<br/>Reported by: Joshua C. Colp<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79899db740484878166b2d7fa8cbdb41389dd99e">[79899db740]</a> George Joseph -- stasis:  Allow filtering by formatter</li>
+</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28104">ASTERISK-28104</a>: AstriCon Feedback:  Automatically create a 1 line dialplan context for stasis apps<br/>Reported by: George Joseph<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1051e1dd1876779f82edfbfffa1115455ca2c269">[1051e1dd18]</a> Ben Ford -- res_stasis: Auto-create context and extens on Stasis app launch.</li>
+</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27959">ASTERISK-27959</a>: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem<br/>Reported by: David Kuehling<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f60afac587d55860c1cc1f6f2fc2e55f1a0ddfc9">[f60afac587]</a> Sean Bright -- res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set</li>
+</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28257">ASTERISK-28257</a>: res_http_websocket: PING / PONG opcodes break data reception<br/>Reported by: Jeremy Lainé<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=907d71b5513d17e61b51798cd924fe97b4a5a3b0">[907d71b551]</a> Jeremy Lainé -- res_http_websocket: ensure control frames do not interfere with data</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28231">ASTERISK-28231</a>: res_http_websocket: Not responding to Connection Close Frame (opcode 8)<br/>Reported by: Jeremy Lainé<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21a1feece286921c47f9d87dc0260a72721ca5a4">[21a1feece2]</a> Jeremy Lainé -- res_http_websocket: respond to CLOSE opcode</li>
+</ul><br><h4>Category: Resources/res_monitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28249">ASTERISK-28249</a>: res_monitor: Segfault with Monitor(wav,file,i)<br/>Reported by: Valentin Vidić<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6506c5b1d46184c6d2749261a048d31232868284">[6506c5b1d4]</a> Valentin Vidic -- channel.c: Fix segfault with Monitor(wav,file,i)</li>
+</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28173">ASTERISK-28173</a>: Deadlock in chan_sip handling subscribe request during res_parking reload<br/>Reported by: Giuseppe Sucameli<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=419db481d1b44091bef52f17cc7acd7957d96c22">[419db481d1]</a> Giuseppe Sucameli -- Fix deadlock handling subscribe req during res_parking reload</li>
+</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28157">ASTERISK-28157</a>: Asterisk crashes when the res_pjsip_* modules unload<br/>Reported by: sungtae kim<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b6df87816bfe0552b7888bf14efd2a82a6c7dbf">[1b6df87816]</a> Sungtae Kim -- res_pjsip: Patch for res_pjsip_* module load/reload crash</li>
+</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28230">ASTERISK-28230</a>: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony<br/>Reported by: David Kuehling<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c6271155fb01700eacddd20651ba9765fabce194">[c6271155fb]</a> Joshua Colp -- res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28162">ASTERISK-28162</a>: [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation<br/>Reported by: Alexei Gradinari<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0e57e458bb309720c1c370abd3aa5088a7d7c17">[c0e57e458b]</a> Alexei Gradinari -- RTP: reset DTMF last seqno/timestamp on RTP renegotiation</li>
+</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28182">ASTERISK-28182</a>: chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE<br/>Reported by: nappsoft<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5de36abd5af0b94a9fab1b8b51c5d1d90a95697a">[5de36abd5a]</a> Pirmin Walthert -- pjproject_bundled: check whether UPDATE is supported on outgoing calls</li>
+</ul><br><h3>Improvement</h3><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28196">ASTERISK-28196</a>: bridge_softmix: Does not support WebRTC source with multi video tracks.<br/>Reported by: Xiemin Chen<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6cf837aede9dd8d805991d1e065baa699145ef2">[f6cf837aed]</a> Xiemin Chen -- bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix</li>
+</ul><br><h4>Category: Formats/format_g726</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28246">ASTERISK-28246</a>: Support skipping on the g726 format<br/>Reported by: Eyal Hasson<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1da2e94a38ed0640b5254f236a8b2e9705a922b">[c1da2e94a3]</a> eyalhasson -- format_g726: add support for seeking</li>
+</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28198">ASTERISK-28198</a>: res_ari: Add new hangup causes for ARI Channel DELETE command<br/>Reported by: Sebastian Damm<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59cf552dd32a55752072439d42461704c64d7167">[59cf552dd3]</a> Sebastian Damm -- res/res_ari: Add additional hangup reasons</li>
+</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28198">ASTERISK-28198</a>: res_ari: Add new hangup causes for ARI Channel DELETE command<br/>Reported by: Sebastian Damm<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59cf552dd32a55752072439d42461704c64d7167">[59cf552dd3]</a> Sebastian Damm -- res/res_ari: Add additional hangup reasons</li>
+</ul><br><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28271">ASTERISK-28271</a>: Opensuse Leap 15 --with-jannson-bundled will not compile<br/>Reported by: David Wilcox<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70fa6e6955ea62f9036f775861cc7425c813c050">[70fa6e6955]</a> George Joseph -- bundled-jansson:  On OpenSuse Leap libjansson.a was placed in lib64</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=19fc99a2fbf899ad5abd7b88b0fc3e225cf7c0ca">19fc99a2fb</a></td><td>sungtae kim</td><td>Added ARI resource /ari/asterisk/ping</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=603143bd5ae2a2321fa2cd7d714a40c922610039">603143bd5a</a></td><td>George Joseph</td><td>media_index.c: Refactored so it doesn't cache the index</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05b79d16ab93b038b39412e2570a21205eb499c4">05b79d16ab</a></td><td>Chris-Savinovich</td><td>Test_cel: Fails when DONT_OPTIMIZE is off</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbef559e0bb24a246558db8b5ddecdd5cf86c857">dbef559e0b</a></td><td>George Joseph</td><td>app_voicemail:  Add Mailbox Aliases</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c11399be3ae48bed620ea5775c435e671495b25">9c11399be3</a></td><td>George Joseph</td><td>pjproject_bundled:  Add patch for double free issue in timer heap</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb6e0df1739b18bae6718e82ab28526054008f01">fb6e0df173</a></td><td>Sean Bright</td><td>pjsip_transport_management: Shutdown transport immediately on disconnect</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=011e46d5a68a986ac503736208507b23dd071868">011e46d5a6</a></td><td>Sean Bright</td><td>sched: Make sched_settime() return void because it cannot fail</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44a862fb576cbc76165f2dc07cc443328b10ea09">44a862fb57</a></td><td>Sean Bright</td><td>res_pjsip_transport_websocket: Don't assert on 0 length payloads</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f22c9f4b7efde0da72c4665cceba380fe1f57b3">7f22c9f4b7</a></td><td>Alexei Gradinari</td><td>res_pjsip: add option to enable ContactStatus event when contact is updated</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f19607870526952d27e6a806e35535e00c0f5f2a">f196078705</a></td><td>Richard Mudgett</td><td>stasic.c: Fix printf format type mismatches with arguments.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59717b5e850f428df58c13b6a24d5baadf0e8c40">59717b5e85</a></td><td>Richard Mudgett</td><td>backtrace.c: Fix casting pointer to/from integral type.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=970805180e300935148fe14cbbf6e35a09666023">970805180e</a></td><td>Sean Bright</td><td>res_rtp_asterisk: Remove some unused structure fields.</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=640aac768bb7208e6baece7860753bbdb2b5564c">640aac768b</a></td><td>Sean Bright</td><td>bridge_builtin_features.c: Set auto(mix)mon variables on both channels</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9febdba05b0767f1938e0261d9f2b5d8f3167902">9febdba05b</a></td><td>Sean Bright</td><td>Use non-blocking socket() and pipe() wrappers</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16ae8330d2955ef2d7bd8e360b700fa9aaf10e1c">16ae8330d2</a></td><td>Sean Bright</td><td>utils: Don't set or clear flags that don't need setting or clearing</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c9519796b9d708a4f9e2fd62053436d22b6e78f">9c9519796b</a></td><td>Sean Bright</td><td>build: Update config.guess and config.sub</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df0b59564e844134ae18be23089ce87aa3f00ca3">df0b59564e</a></td><td>George Joseph</td><td>Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8a18fb81c1a4885b9b6568d69e56ab40f488d02a">8a18fb81c1</a></td><td>Sean Bright</td><td>utils: Wrap socket() and pipe() to reduce syscalls</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1657508ddd1e204dddc940876810d40c575066e8">1657508ddd</a></td><td>David M. Lee</td><td>Removing registrar_expire from basic-pbx config</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6c2662404e6094b8554b49bab672dfc47e9b8c6">a6c2662404</a></td><td>George Joseph</td><td>CI: Various updates to buildAsterisk.sh</td></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60e548ffa5f0eefca1ba23295ae01b97942a5b5c">60e548ffa5</a></td><td>Chris-Savinovich</td><td>test_websocket_client.c: Disable websocket_client_create_and_connect test.</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>.lastclean                                                                                       |    1
+.version                                                                                         |    1
+ChangeLog                                                                                        |81923 ----------
+asterisk-16.1.0-summary.html                                                                     |  620
+asterisk-16.1.0-summary.txt                                                                      | 1442
+b/CHANGES                                                                                        |   49
+b/apps/app_confbridge.c                                                                          |    2
+b/apps/app_queue.c                                                                               |    2
+b/apps/app_voicemail.c                                                                           |  335
+b/apps/confbridge/conf_state_empty.c                                                             |    3
+b/apps/confbridge/conf_state_inactive.c                                                          |    2
+b/apps/confbridge/include/confbridge.h                                                           |    8
+b/bridges/bridge_builtin_features.c                                                              |    2
+b/bridges/bridge_softmix.c                                                                       |   16
+b/channels/chan_sip.c                                                                            |    6
+b/config.guess                                                                                   |  666
+b/config.sub                                                                                     | 2535
+b/configs/basic-pbx/modules.conf                                                                 |    1
+b/configs/samples/codecs.conf.sample                                                             |   26
+b/configs/samples/pjsip.conf.sample                                                              |    5
+b/configs/samples/voicemail.conf.sample                                                          |   12
+b/configure                                                                                      |   86
+b/configure.ac                                                                                   |   28
+b/contrib/ast-db-manage/config/versions/0838f8db6a61_pjsip_add_send_contact_status_on_update_.py |   39
+b/contrib/scripts/ast_coredumper                                                                 |  111
+b/formats/format_g726.c                                                                          |   35
+b/include/asterisk/autoconfig.h.in                                                               |    6
+b/include/asterisk/channel.h                                                                     |   12
+b/include/asterisk/media_index.h                                                                 |   20
+b/include/asterisk/res_pjsip.h                                                                   |    9
+b/include/asterisk/res_pjsip_session.h                                                           |   13
+b/include/asterisk/sounds_index.h                                                                |   13
+b/include/asterisk/stasis.h                                                                      |   51
+b/include/asterisk/stasis_internal.h                                                             |    5
+b/include/asterisk/stasis_message_router.h                                                       |   54
+b/include/asterisk/strings.h                                                                     |   14
+b/include/asterisk/utils.h                                                                       |   42
+b/main/alertpipe.c                                                                               |   11
+b/main/asterisk.c                                                                                |    4
+b/main/asterisk.exports.in                                                                       |    1
+b/main/backtrace.c                                                                               |   10
+b/main/channel.c                                                                                 |   10
+b/main/channel_internal_api.c                                                                    |   12
+b/main/manager.c                                                                                 |    4
+b/main/manager_channels.c                                                                        |   10
+b/main/media_index.c                                                                             |  229
+b/main/pbx.c                                                                                     |   85
+b/main/sched.c                                                                                   |   20
+b/main/sounds.c                                                                                  |  179
+b/main/stasis.c                                                                                  |  877
+b/main/stasis_cache.c                                                                            |   33
+b/main/stasis_message.c                                                                          |   16
+b/main/stasis_message_router.c                                                                   |   71
+b/main/strings.c                                                                                 |   15
+b/main/tcptls.c                                                                                  |    3
+b/main/udptl.c                                                                                   |    3
+b/main/utils.c                                                                                   |   44
+b/res/ari/ari_model_validators.c                                                                 |   70
+b/res/ari/ari_model_validators.h                                                                 |   22
+b/res/ari/resource_asterisk.c                                                                    |   18
+b/res/ari/resource_asterisk.h                                                                    |   11
+b/res/ari/resource_channels.c                                                                    |   16
+b/res/ari/resource_sounds.c                                                                      |   28
+b/res/res_agi.c                                                                                  |    7
+b/res/res_ari_asterisk.c                                                                         |   63
+b/res/res_format_attr_h264.c                                                                     |    2
+b/res/res_http_websocket.c                                                                       |   50
+b/res/res_pjsip.c                                                                                |    3
+b/res/res_pjsip/config_global.c                                                                  |   72
+b/res/res_pjsip/include/res_pjsip_private.h                                                      |   10
+b/res/res_pjsip/pjsip_configuration.c                                                            |   35
+b/res/res_pjsip/pjsip_message_filter.c                                                           |    1
+b/res/res_pjsip/pjsip_options.c                                                                  |   55
+b/res/res_pjsip/pjsip_session.c                                                                  |   85
+b/res/res_pjsip/pjsip_transport_management.c                                                     |   77
+b/res/res_pjsip_registrar.c                                                                      |   29
+b/res/res_pjsip_sdp_rtp.c                                                                        |    8
+b/res/res_pjsip_session.c                                                                        |   68
+b/res/res_pjsip_transport_websocket.c                                                            |   13
+b/res/res_rtp_asterisk.c                                                                         |   37
+b/res/res_timing_pthread.c                                                                       |    7
+b/res/stasis/app.c                                                                               |   51
+b/rest-api/api-docs/asterisk.json                                                                |   33
+b/rest-api/api-docs/channels.json                                                                |    8
+b/tests/CI/buildAsterisk.sh                                                                      |  163
+b/tests/test_stasis.c                                                                            |  397
+b/tests/test_websocket_client.c                                                                  |    1
+b/third-party/jansson/Makefile                                                                   |    3
+b/third-party/jansson/configure.m4                                                               |    4
+b/third-party/pjproject/configure.m4                                                             |    4
+b/third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch                 |   19
+contrib/realtime/mssql/mssql_cdr.sql                                                             |   59
+contrib/realtime/mssql/mssql_config.sql                                                          | 2118
+contrib/realtime/mssql/mssql_voicemail.sql                                                       |   55
+contrib/realtime/mysql/mysql_cdr.sql                                                             |   41
+contrib/realtime/mysql/mysql_config.sql                                                          | 1213
+contrib/realtime/mysql/mysql_voicemail.sql                                                       |   35
+contrib/realtime/oracle/oracle_cdr.sql                                                           |   53
+contrib/realtime/oracle/oracle_config.sql                                                        | 2076
+contrib/realtime/oracle/oracle_voicemail.sql                                                     |   49
+contrib/realtime/postgresql/postgresql_cdr.sql                                                   |   45
+contrib/realtime/postgresql/postgresql_config.sql                                                | 1309
+contrib/realtime/postgresql/postgresql_voicemail.sql                                             |   39
+103 files changed, 5028 insertions(+), 93366 deletions(-)</pre><br></html>
--- a/asterisk-16.2.0-rc1-summary.txt
+++ b/asterisk-16.2.0-rc1-summary.txt
@@ -0,0 +1,533 @@
+                                Release Summary
+
+                              asterisk-16.2.0-rc1
+
+                                Date: 2019-02-06
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Open Issues
+    5. Other Changes
+    6. 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-16.1.0.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were affected by commits that went into
+   this release.
+
+   Coders                   Testers                  Reporters                
+   10 George Joseph                                  4 Joshua C. Colp         
+   10 Sean Bright                                    3 George Joseph          
+   3 Kevin Harwell                                   2 Alexei Gradinari       
+   3 Alexei Gradinari                                2 Giuseppe Sucameli      
+   2 Joshua C. Colp                                  2 Jeremy LainÃ©          
+   2 Jeremy LainÃ©                                   2 David Kuehling         
+   2 Giuseppe Sucameli                               1 Jean Aunis - Prescom   
+   2 Joshua Colp                                     1 Andrew Nagy            
+   2 Chris-Savinovich                                1 boatright              
+   2 Richard Mudgett                                 1 Mohit Dhiman           
+   1 Xiemin Chen                                     1 sungtae kim            
+   1 Mohit Dhiman                                    1 Ray                    
+   1 Pirmin Walthert                                 1 Eyal Hasson            
+   1 Sungtae Kim                                     1 Ross Beer              
+   1 Diederik de Groot                               1 abelbeck               
+   1 David M. Lee                                    1 nappsoft               
+   1 Jean Aunis                                      1 Gianluca Merlo         
+   1 Corey Farrell                                   1 Xiemin Chen            
+   1 Bryan Boatright                                 1 David Wilcox           
+   1 Valentin Vidic                                  1 Andrew Nagy            
+   1 sungtae kim                                     1 Mark                   
+   1 Gerald Schnabel                                 1 Diederik de Groot      
+   1 Ben Ford                                        1 Valentin VidiÄ*        
+   1 eyalhasson                                      1 Gerald Schnabel        
+   1 Sebastian Damm                                  1 xiemchen               
+                                                     1 David Wilcox           
+                                                     1 Sebastian Damm         
+                                                     1 David Kuehling         
+
+     ----------------------------------------------------------------------
+
+                                 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.
+
+  Bug
+
+    Category: . I did not set the category correctly.
+
+   ASTERISK-28221: Bug in ast_coredumper
+   Reported by: Andrew Nagy
+     * [3efe5061d5] George Joseph -- ast_coredumper: Refactor the pid
+       determination process
+
+    Category: Applications/app_confbridge
+
+   ASTERISK-28201: [patch] confbridge: no announce to the marked users when
+   they join an empty conference
+   Reported by: Alexei Gradinari
+     * [2610379605] Alexei Gradinari -- confbridge: announce to the marked
+       users when they join an empty conference
+
+    Category: Applications/app_queue
+
+   ASTERISK-28218: app_queue: Asterisk crashes when using Queue with a
+   pre-dial handler (option b)
+   Reported by: Mark
+     * [2d9482695d] Joshua Colp -- app_queue: Fix crash when using 'b' option
+       on non-ringall queue.
+
+    Category: Applications/app_voicemail
+
+   ASTERISK-28225: app_voicemail: Channel variable VM_MESSAGEFILE not updated
+   correctly if message marked "urgent"
+   Reported by: boatright
+     * [92298434bd] Bryan Boatright -- app_voicemail: Fix Channel variable
+       VM_MESSAGEFILE for "urgent" voicemail
+   ASTERISK-28222: Regression: MWI polling no longer works
+   Reported by: abelbeck
+     * [ff2ed4eeee] George Joseph -- Revert "stasis_cache: Stop caching
+       stasis subscription change messages"
+   ASTERISK-28215: app_voicemail: Leaving voicemail sometimes doesn't trigger
+   NOTIFYs
+   Reported by: George Joseph
+     * [aebb822d1f] George Joseph -- app_voicemail: Don't delete mailbox
+       state unless mailbox is deleted
+
+    Category: Channels/chan_pjsip
+
+   ASTERISK-28238: PJSIP realtime. getcontext not working with DUNDI
+   Reported by: Ray
+     * [9c3b4dcf80] Kevin Harwell -- pjsip/config_global: regcontext context
+       not created
+   ASTERISK-28213: res_pjsip: Threads pile up needlessly when AOR is blocked
+   Reported by: Ross Beer
+     * [f1fb249132] Kevin Harwell -- res_pjsip_registrar: mitigate blocked
+       threads on reliable transport shutdown
+   ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite,
+   we're not trying UPDATE
+   Reported by: George Joseph
+     * [5de36abd5a] Pirmin Walthert -- pjproject_bundled: check whether
+       UPDATE is supported on outgoing calls
+
+    Category: Channels/chan_sip/General
+
+   ASTERISK-28194: chan_sip: Leak using contact ACL
+   Reported by: Giuseppe Sucameli
+     * [6071ad77f5] Giuseppe Sucameli -- chan_sip: Fix leak using contact ACL
+
+    Category: Channels/chan_sip/Subscriptions
+
+   ASTERISK-28173: Deadlock in chan_sip handling subscribe request during
+   res_parking reload
+   Reported by: Giuseppe Sucameli
+     * [419db481d1] Giuseppe Sucameli -- Fix deadlock handling subscribe req
+       during res_parking reload
+
+    Category: Codecs/codec_opus
+
+   ASTERISK-28263: codec_opus: errors setting max_playback_rate and bitrate
+   to "sdp"
+   Reported by: Gianluca Merlo
+     * [f6452f9656] Kevin Harwell -- codecs.conf.sample: update codec opus
+       docs
+
+    Category: Core/BuildSystem
+
+   ASTERISK-28250: build: Cross-compilation fails for target
+   arm-linux-gnueabihf
+   Reported by: Jean Aunis - Prescom
+     * [d3a6714158] Jean Aunis -- build : Fix cross-compilation errors
+
+    Category: Core/Channels
+
+   ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of
+   channels past destruction in certain cases
+   Reported by: Mohit Dhiman
+     * [4b24da607e] Mohit Dhiman -- stasis/endpoint: Fix memory leak of
+       channel_ids in ast_endpoint structure.
+
+    Category: Core/General
+
+   ASTERISK-28232: core: RAII using clang use-after-scope issue
+   Reported by: Diederik de Groot
+     * [d2c182b6ab] Diederik de Groot -- RAII: Change order or variables in
+       clang version
+
+    Category: Core/Stasis
+
+   ASTERISK-28252: HangupHandler manager events are never thrown
+   Reported by: Gerald Schnabel
+     * [735bd4d185] Gerald Schnabel -- manager_channels: Fix throwing of
+       HangupHandler manager events
+   ASTERISK-28244: stasis: Filter messages at publishing to AMI/ARI
+   Reported by: Joshua C. Colp
+     * [fcd07c34fb] Joshua C. Colp -- stasis / manager / ari: Better filter
+       messages.
+   ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of
+   channels past destruction in certain cases
+   Reported by: Mohit Dhiman
+     * [4b24da607e] Mohit Dhiman -- stasis/endpoint: Fix memory leak of
+       channel_ids in ast_endpoint structure.
+   ASTERISK-28212: stasis: Statistics broke ABI under developer mode
+   Reported by: Joshua C. Colp
+     * [44a7faca21] Corey Farrell -- stasis: Fix ABI between DEVMODE and
+       non-DEVMODE.
+   ASTERISK-28117: stasis: Add statistics for usage when in developer mode
+   Reported by: Joshua C. Colp
+     * [68ec7d93e8] Joshua C. Colp -- stasis: Add statistics gathering in
+       developer mode.
+   ASTERISK-28186: stasis: Filter messages at publishing based on to_*
+   presence
+   Reported by: Joshua C. Colp
+     * [79899db740] George Joseph -- stasis: Allow filtering by formatter
+
+    Category: Resources/res_ari
+
+   ASTERISK-28104: AstriCon Feedback: Automatically create a 1 line dialplan
+   context for stasis apps
+   Reported by: George Joseph
+     * [1051e1dd18] Ben Ford -- res_stasis: Auto-create context and extens on
+       Stasis app launch.
+
+    Category: Resources/res_format_attr_h264
+
+   ASTERISK-27959: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
+   Reported by: David Kuehling
+     * [f60afac587] Sean Bright -- res_format_attr_h264.c: Make sure
+       profile-level-id fmtp attribute is set
+
+    Category: Resources/res_http_websocket
+
+   ASTERISK-28257: res_http_websocket: PING / PONG opcodes break data
+   reception
+   Reported by: Jeremy LainÃ©
+     * [907d71b551] Jeremy LainÃ© -- res_http_websocket: ensure control
+       frames do not interfere with data
+   ASTERISK-28231: res_http_websocket: Not responding to Connection Close
+   Frame (opcode 8)
+   Reported by: Jeremy LainÃ©
+     * [21a1feece2] Jeremy LainÃ© -- res_http_websocket: respond to CLOSE
+       opcode
+
+    Category: Resources/res_monitor
+
+   ASTERISK-28249: res_monitor: Segfault with Monitor(wav,file,i)
+   Reported by: Valentin VidiÄ*
+     * [6506c5b1d4] Valentin Vidic -- channel.c: Fix segfault with
+       Monitor(wav,file,i)
+
+    Category: Resources/res_parking
+
+   ASTERISK-28173: Deadlock in chan_sip handling subscribe request during
+   res_parking reload
+   Reported by: Giuseppe Sucameli
+     * [419db481d1] Giuseppe Sucameli -- Fix deadlock handling subscribe req
+       during res_parking reload
+
+    Category: Resources/res_pjsip_session
+
+   ASTERISK-28157: Asterisk crashes when the res_pjsip_* modules unload
+   Reported by: sungtae kim
+     * [1b6df87816] Sungtae Kim -- res_pjsip: Patch for res_pjsip_* module
+       load/reload crash
+
+    Category: Resources/res_rtp_asterisk
+
+   ASTERISK-28230: res_rtp_asterisk: abs-send-time extension added with
+   Asterisk 15.5.0 breaks GXV3140 video telephony
+   Reported by: David Kuehling
+     * [c6271155fb] Joshua Colp -- res_pjsip_sdp_rtp: Only enable
+       abs-send-time when WebRTC is enabled.
+   ASTERISK-28162: [patch] need to reset DTMF last sequence number and
+   timestamp on RTP renegotiation
+   Reported by: Alexei Gradinari
+     * [c0e57e458b] Alexei Gradinari -- RTP: reset DTMF last seqno/timestamp
+       on RTP renegotiation
+
+    Category: Third-Party/pjproject
+
+   ASTERISK-28182: chan_pjsip: When connected_line_method is set to invite,
+   asterisk is not trying UPDATE
+   Reported by: nappsoft
+     * [5de36abd5a] Pirmin Walthert -- pjproject_bundled: check whether
+       UPDATE is supported on outgoing calls
+
+  Improvement
+
+    Category: Bridges/bridge_softmix
+
+   ASTERISK-28196: bridge_softmix: Does not support WebRTC source with multi
+   video tracks.
+   Reported by: Xiemin Chen
+     * [f6cf837aed] Xiemin Chen -- bridge_softmix: Use MSID:LABEL metadata as
+       the cloned stream's appendix
+
+    Category: Formats/format_g726
+
+   ASTERISK-28246: Support skipping on the g726 format
+   Reported by: Eyal Hasson
+     * [c1da2e94a3] eyalhasson -- format_g726: add support for seeking
+
+    Category: Resources/res_ari
+
+   ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE
+   command
+   Reported by: Sebastian Damm
+     * [59cf552dd3] Sebastian Damm -- res/res_ari: Add additional hangup
+       reasons
+
+    Category: Resources/res_ari_channels
+
+   ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE
+   command
+   Reported by: Sebastian Damm
+     * [59cf552dd3] Sebastian Damm -- res/res_ari: Add additional hangup
+       reasons
+
+     ----------------------------------------------------------------------
+
+                                  Open Issues
+
+                                 [Back to Top]
+
+   This is a list of all open issues from the issue tracker that were
+   referenced by changes that went into this release.
+
+  Bug
+
+    Category: Core/BuildSystem
+
+   ASTERISK-28271: Opensuse Leap 15 --with-jannson-bundled will not compile
+   Reported by: David Wilcox
+     * [70fa6e6955] George Joseph -- bundled-jansson: On OpenSuse Leap
+       libjansson.a was placed in lib64
+
+     ----------------------------------------------------------------------
+
+                      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                                |
+   |------------+------------------+----------------------------------------|
+   | 19fc99a2fb | sungtae kim      | Added ARI resource /ari/asterisk/ping  |
+   |------------+------------------+----------------------------------------|
+   | 603143bd5a | George Joseph    | media_index.c: Refactored so it        |
+   |            |                  | doesn't cache the index                |
+   |------------+------------------+----------------------------------------|
+   | 05b79d16ab | Chris-Savinovich | Test_cel: Fails when DONT_OPTIMIZE is  |
+   |            |                  | off                                    |
+   |------------+------------------+----------------------------------------|
+   | dbef559e0b | George Joseph    | app_voicemail: Add Mailbox Aliases     |
+   |------------+------------------+----------------------------------------|
+   | 9c11399be3 | George Joseph    | pjproject_bundled: Add patch for       |
+   |            |                  | double free issue in timer heap        |
+   |------------+------------------+----------------------------------------|
+   | fb6e0df173 | Sean Bright      | pjsip_transport_management: Shutdown   |
+   |            |                  | transport immediately on disconnect    |
+   |------------+------------------+----------------------------------------|
+   | 011e46d5a6 | Sean Bright      | sched: Make sched_settime() return     |
+   |            |                  | void because it cannot fail            |
+   |------------+------------------+----------------------------------------|
+   | 44a862fb57 | Sean Bright      | res_pjsip_transport_websocket: Don't   |
+   |            |                  | assert on 0 length payloads            |
+   |------------+------------------+----------------------------------------|
+   |            |                  | res_pjsip: add option to enable        |
+   | 7f22c9f4b7 | Alexei Gradinari | ContactStatus event when contact is    |
+   |            |                  | updated                                |
+   |------------+------------------+----------------------------------------|
+   | f196078705 | Richard Mudgett  | stasic.c: Fix printf format type       |
+   |            |                  | mismatches with arguments.             |
+   |------------+------------------+----------------------------------------|
+   | 59717b5e85 | Richard Mudgett  | backtrace.c: Fix casting pointer       |
+   |            |                  | to/from integral type.                 |
+   |------------+------------------+----------------------------------------|
+   | 970805180e | Sean Bright      | res_rtp_asterisk: Remove some unused   |
+   |            |                  | structure fields.                      |
+   |------------+------------------+----------------------------------------|
+   |            |                  | bridge_builtin_features.c: Set         |
+   | 640aac768b | Sean Bright      | auto(mix)mon variables on both         |
+   |            |                  | channels                               |
+   |------------+------------------+----------------------------------------|
+   | 9febdba05b | Sean Bright      | Use non-blocking socket() and pipe()   |
+   |            |                  | wrappers                               |
+   |------------+------------------+----------------------------------------|
+   | 16ae8330d2 | Sean Bright      | utils: Don't set or clear flags that   |
+   |            |                  | don't need setting or clearing         |
+   |------------+------------------+----------------------------------------|
+   | 9c9519796b | Sean Bright      | build: Update config.guess and         |
+   |            |                  | config.sub                             |
+   |------------+------------------+----------------------------------------|
+   |            |                  | Revert "RTP: reset DTMF last           |
+   | df0b59564e | George Joseph    | seqno/timestamp on voice packet with   |
+   |            |                  | marker bit"                            |
+   |------------+------------------+----------------------------------------|
+   | 8a18fb81c1 | Sean Bright      | utils: Wrap socket() and pipe() to     |
+   |            |                  | reduce syscalls                        |
+   |------------+------------------+----------------------------------------|
+   | 1657508ddd | David M. Lee     | Removing registrar_expire from         |
+   |            |                  | basic-pbx config                       |
+   |------------+------------------+----------------------------------------|
+   | a6c2662404 | George Joseph    | CI: Various updates to                 |
+   |            |                  | buildAsterisk.sh                       |
+   |------------+------------------+----------------------------------------|
+   |            |                  | test_websocket_client.c: Disable       |
+   | 60e548ffa5 | Chris-Savinovich | websocket_client_create_and_connect    |
+   |            |                  | test.                                  |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                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.
+
+ .lastclean                                                                                       |    1
+ .version                                                                                         |    1
+ ChangeLog                                                                                        |81923 ----------
+ asterisk-16.1.0-summary.html                                                                     |  620
+ asterisk-16.1.0-summary.txt                                                                      | 1442
+ b/CHANGES                                                                                        |   49
+ b/apps/app_confbridge.c                                                                          |    2
+ b/apps/app_queue.c                                                                               |    2
+ b/apps/app_voicemail.c                                                                           |  335
+ b/apps/confbridge/conf_state_empty.c                                                             |    3
+ b/apps/confbridge/conf_state_inactive.c                                                          |    2
+ b/apps/confbridge/include/confbridge.h                                                           |    8
+ b/bridges/bridge_builtin_features.c                                                              |    2
+ b/bridges/bridge_softmix.c                                                                       |   16
+ b/channels/chan_sip.c                                                                            |    6
+ b/config.guess                                                                                   |  666
+ b/config.sub                                                                                     | 2535
+ b/configs/basic-pbx/modules.conf                                                                 |    1
+ b/configs/samples/codecs.conf.sample                                                             |   26
+ b/configs/samples/pjsip.conf.sample                                                              |    5
+ b/configs/samples/voicemail.conf.sample                                                          |   12
+ b/configure                                                                                      |   86
+ b/configure.ac                                                                                   |   28
+ b/contrib/ast-db-manage/config/versions/0838f8db6a61_pjsip_add_send_contact_status_on_update_.py |   39
+ b/contrib/scripts/ast_coredumper                                                                 |  111
+ b/formats/format_g726.c                                                                          |   35
+ b/include/asterisk/autoconfig.h.in                                                               |    6
+ b/include/asterisk/channel.h                                                                     |   12
+ b/include/asterisk/media_index.h                                                                 |   20
+ b/include/asterisk/res_pjsip.h                                                                   |    9
+ b/include/asterisk/res_pjsip_session.h                                                           |   13
+ b/include/asterisk/sounds_index.h                                                                |   13
+ b/include/asterisk/stasis.h                                                                      |   51
+ b/include/asterisk/stasis_internal.h                                                             |    5
+ b/include/asterisk/stasis_message_router.h                                                       |   54
+ b/include/asterisk/strings.h                                                                     |   14
+ b/include/asterisk/utils.h                                                                       |   42
+ b/main/alertpipe.c                                                                               |   11
+ b/main/asterisk.c                                                                                |    4
+ b/main/asterisk.exports.in                                                                       |    1
+ b/main/backtrace.c                                                                               |   10
+ b/main/channel.c                                                                                 |   10
+ b/main/channel_internal_api.c                                                                    |   12
+ b/main/manager.c                                                                                 |    4
+ b/main/manager_channels.c                                                                        |   10
+ b/main/media_index.c                                                                             |  229
+ b/main/pbx.c                                                                                     |   85
+ b/main/sched.c                                                                                   |   20
+ b/main/sounds.c                                                                                  |  179
+ b/main/stasis.c                                                                                  |  877
+ b/main/stasis_cache.c                                                                            |   33
+ b/main/stasis_message.c                                                                          |   16
+ b/main/stasis_message_router.c                                                                   |   71
+ b/main/strings.c                                                                                 |   15
+ b/main/tcptls.c                                                                                  |    3
+ b/main/udptl.c                                                                                   |    3
+ b/main/utils.c                                                                                   |   44
+ b/res/ari/ari_model_validators.c                                                                 |   70
+ b/res/ari/ari_model_validators.h                                                                 |   22
+ b/res/ari/resource_asterisk.c                                                                    |   18
+ b/res/ari/resource_asterisk.h                                                                    |   11
+ b/res/ari/resource_channels.c                                                                    |   16
+ b/res/ari/resource_sounds.c                                                                      |   28
+ b/res/res_agi.c                                                                                  |    7
+ b/res/res_ari_asterisk.c                                                                         |   63
+ b/res/res_format_attr_h264.c                                                                     |    2
+ b/res/res_http_websocket.c                                                                       |   50
+ b/res/res_pjsip.c                                                                                |    3
+ b/res/res_pjsip/config_global.c                                                                  |   72
+ b/res/res_pjsip/include/res_pjsip_private.h                                                      |   10
+ b/res/res_pjsip/pjsip_configuration.c                                                            |   35
+ b/res/res_pjsip/pjsip_message_filter.c                                                           |    1
+ b/res/res_pjsip/pjsip_options.c                                                                  |   55
+ b/res/res_pjsip/pjsip_session.c                                                                  |   85
+ b/res/res_pjsip/pjsip_transport_management.c                                                     |   77
+ b/res/res_pjsip_registrar.c                                                                      |   29
+ b/res/res_pjsip_sdp_rtp.c                                                                        |    8
+ b/res/res_pjsip_session.c                                                                        |   68
+ b/res/res_pjsip_transport_websocket.c                                                            |   13
+ b/res/res_rtp_asterisk.c                                                                         |   37
+ b/res/res_timing_pthread.c                                                                       |    7
+ b/res/stasis/app.c                                                                               |   51
+ b/rest-api/api-docs/asterisk.json                                                                |   33
+ b/rest-api/api-docs/channels.json                                                                |    8
+ b/tests/CI/buildAsterisk.sh                                                                      |  163
+ b/tests/test_stasis.c                                                                            |  397
+ b/tests/test_websocket_client.c                                                                  |    1
+ b/third-party/jansson/Makefile                                                                   |    3
+ b/third-party/jansson/configure.m4                                                               |    4
+ b/third-party/pjproject/configure.m4                                                             |    4
+ b/third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch                 |   19
+ contrib/realtime/mssql/mssql_cdr.sql                                                             |   59
+ contrib/realtime/mssql/mssql_config.sql                                                          | 2118
+ contrib/realtime/mssql/mssql_voicemail.sql                                                       |   55
+ contrib/realtime/mysql/mysql_cdr.sql                                                             |   41
+ contrib/realtime/mysql/mysql_config.sql                                                          | 1213
+ contrib/realtime/mysql/mysql_voicemail.sql                                                       |   35
+ contrib/realtime/oracle/oracle_cdr.sql                                                           |   53
+ contrib/realtime/oracle/oracle_config.sql                                                        | 2076
+ contrib/realtime/oracle/oracle_voicemail.sql                                                     |   49
+ contrib/realtime/postgresql/postgresql_cdr.sql                                                   |   45
+ contrib/realtime/postgresql/postgresql_config.sql                                                | 1309
+ contrib/realtime/postgresql/postgresql_voicemail.sql                                             |   39
+ 103 files changed, 5028 insertions(+), 93366 deletions(-)
--- a/contrib/realtime/mssql/mssql_cdr.sql
+++ b/contrib/realtime/mssql/mssql_cdr.sql
@@ -0,0 +1,59 @@
+BEGIN TRANSACTION;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+GO
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20) NULL, 
+    src VARCHAR(80) NULL, 
+    dst VARCHAR(80) NULL, 
+    dcontext VARCHAR(80) NULL, 
+    clid VARCHAR(80) NULL, 
+    channel VARCHAR(80) NULL, 
+    dstchannel VARCHAR(80) NULL, 
+    lastapp VARCHAR(80) NULL, 
+    lastdata VARCHAR(80) NULL, 
+    start DATETIME NULL, 
+    answer DATETIME NULL, 
+    [end] DATETIME NULL, 
+    duration INTEGER NULL, 
+    billsec INTEGER NULL, 
+    disposition VARCHAR(45) NULL, 
+    amaflags VARCHAR(45) NULL, 
+    userfield VARCHAR(256) NULL, 
+    uniqueid VARCHAR(150) NULL, 
+    linkedid VARCHAR(150) NULL, 
+    peeraccount VARCHAR(20) NULL, 
+    sequence INTEGER NULL
+);
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+GO
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr ALTER COLUMN accountcode VARCHAR(80);
+
+GO
+
+ALTER TABLE cdr ALTER COLUMN peeraccount VARCHAR(80);
+
+GO
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+
+GO
+
+COMMIT;
+
+GO
+
--- a/contrib/realtime/mssql/mssql_config.sql
+++ b/contrib/realtime/mssql/mssql_config.sql
--- a/contrib/realtime/mssql/mssql_voicemail.sql
+++ b/contrib/realtime/mssql/mssql_voicemail.sql
@@ -0,0 +1,55 @@
+BEGIN TRANSACTION;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+GO
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80) NULL, 
+    macrocontext VARCHAR(80) NULL, 
+    callerid VARCHAR(80) NULL, 
+    origtime INTEGER NULL, 
+    duration INTEGER NULL, 
+    recording IMAGE NULL, 
+    flag VARCHAR(30) NULL, 
+    category VARCHAR(30) NULL, 
+    mailboxuser VARCHAR(30) NULL, 
+    mailboxcontext VARCHAR(30) NULL, 
+    msg_id VARCHAR(40) NULL
+);
+
+GO
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+GO
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+GO
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+GO
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE;
+
+GO
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+GO
+
+COMMIT;
+
+GO
+
--- 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/oracle/oracle_cdr.sql
+++ b/contrib/realtime/oracle/oracle_cdr.sql
@@ -0,0 +1,53 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR2(32 CHAR) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+)
+
+/
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR2(20 CHAR), 
+    src VARCHAR2(80 CHAR), 
+    dst VARCHAR2(80 CHAR), 
+    dcontext VARCHAR2(80 CHAR), 
+    clid VARCHAR2(80 CHAR), 
+    channel VARCHAR2(80 CHAR), 
+    dstchannel VARCHAR2(80 CHAR), 
+    lastapp VARCHAR2(80 CHAR), 
+    lastdata VARCHAR2(80 CHAR), 
+    "start" DATE, 
+    answer DATE, 
+    end DATE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR2(45 CHAR), 
+    amaflags VARCHAR2(45 CHAR), 
+    userfield VARCHAR2(256 CHAR), 
+    uniqueid VARCHAR2(150 CHAR), 
+    linkedid VARCHAR2(150 CHAR), 
+    peeraccount VARCHAR2(20 CHAR), 
+    sequence INTEGER
+)
+
+/
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d')
+
+/
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr MODIFY accountcode VARCHAR2(80 CHAR)
+
+/
+
+ALTER TABLE cdr MODIFY peeraccount VARCHAR2(80 CHAR)
+
+/
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d'
+
+/
+
--- a/contrib/realtime/oracle/oracle_config.sql
+++ b/contrib/realtime/oracle/oracle_config.sql
--- a/contrib/realtime/oracle/oracle_voicemail.sql
+++ b/contrib/realtime/oracle/oracle_voicemail.sql
@@ -0,0 +1,49 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR2(32 CHAR) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+)
+
+/
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR2(255 CHAR) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR2(80 CHAR), 
+    macrocontext VARCHAR2(80 CHAR), 
+    callerid VARCHAR2(80 CHAR), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR2(30 CHAR), 
+    category VARCHAR2(30 CHAR), 
+    mailboxuser VARCHAR2(30 CHAR), 
+    mailboxcontext VARCHAR2(30 CHAR), 
+    msg_id VARCHAR2(40 CHAR)
+)
+
+/
+
+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
+
+/
+
+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;
+