]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Release summaries: Remove previous versions
authorKevin Harwell <kharwell@lunkwill.digium.internal>
Wed, 7 Oct 2015 18:14:19 +0000 (13:14 -0500)
committerKevin Harwell <kharwell@lunkwill.digium.internal>
Wed, 7 Oct 2015 18:14:19 +0000 (13:14 -0500)
asterisk-11.20.0-rc1-summary.html [deleted file]
asterisk-11.20.0-rc1-summary.txt [deleted file]

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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-11.20.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-11.20.0-rc1</h3><h3 align="center">Date: 2015-09-29</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-11.19.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
-<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
-<tr valign="top"><td width="33%">9 Richard Mudgett <rmudgett@digium.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 David M. Lee <dlee@respoke.io><br/>2 Kevin Harwell <kharwell@digium.com><br/>2 Mark Michelson <mmichelson@digium.com><br/>2 Kevin Harwell <kharwell@lunkwill><br/>2 Alexander Anikin <may213@yandex.ru><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Mark Duncan <mark@syon.co.jp><br/>1 Scott Griepentrog <scott@griepentrog.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Alexander Traud <pabstraud@compuserve.com><br/></td><td width="33%">1 Elazar Broad<br/></td><td width="33%">3 Joshua Colp <jcolp@digium.com><br/>2 Stefan Engström <stefanen@kth.se><br/>2 Richard Mudgett <rmudgett@digium.com><br/>2 John Hardin<br/>2 Alexandr Dranchuk <alex.dranchuk@gmail.com><br/>2 Kevin Harwell <kharwell@digium.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Kevin Harwell<br/>1 Ivan Poddubny<br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Elazar Broad<br/>1 Lorne Gaetz<br/>1 Scott Griepentrog <sgriepentrog@digium.com><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Guido Falsi<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Alexandr Dranchuk<br/>1 Alexander Traud <pabstraud@compuserve.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>Bug</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25227">ASTERISK-25227</a>: No audio at in-band announcements in ooh323 channel<br/>Reported by: Alexandr Dranchuk<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=198a1cab8ed070fd077361ca931898850e7b0d49">[198a1cab8e]</a> Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE with inband info available</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25299">ASTERISK-25299</a>: RTP port leaks with incoming OOH323 calls<br/>Reported by: Alexandr Dranchuk<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07b25a2312efecdfc4aeccaf8742782e714ff358">[07b25a2312]</a> Alexander Anikin -- chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy</li>
-</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25215">ASTERISK-25215</a>: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember<br/>Reported by: Lorne Gaetz<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43150cc58d88a79017f930edff2ae8d62bb9bb79">[43150cc58d]</a> Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.</li>
-</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25410">ASTERISK-25410</a>: app_record: RECORDED_FILE variable not being populated<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=455a31476bfe26a0cf3fdb186cdf9f0d6f0696ee">[455a31476b]</a> Kevin Harwell -- app_record: RECORDED_FILE variable not being populated</li>
-</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25315">ASTERISK-25315</a>: DAHDI channels send shortened duration DTMF tones.<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c777c9565dbcb8527f5a9b2dbc35e81c541101f1">[c777c9565d]</a> Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f43ea74e9eb18eaead46a08ac287ee5ce0ad798a">[f43ea74e9e]</a> Richard Mudgett -- chan_dahdi.c: Lock private struct for ast_write().</li>
-</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25346">ASTERISK-25346</a>: chan_sip: Overwriting answered elsewhere hangup cause on call pickup<br/>Reported by: Joshua Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59636e82b2140993081837b0719bf6ad0ec41c40">[59636e82b2]</a> Joshua Colp -- chan_sip: Allow call pickup to set the hangup cause.</li>
-</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25396">ASTERISK-25396</a>: chan_sip: Extremely long callerid name causes invalid SIP<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fcf45e9cfc3a68ed4b37ba8cafdf73279f5aae7">[2fcf45e9cf]</a> Walter Doekes -- chan_sip: Fix From header truncation for extremely long CALLERID(name).</li>
-</ul><br><h4>Category: Channels/chan_sip/Security Framework</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25320">ASTERISK-25320</a>: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=430db4333e25d4d1c101ed4a3473af320abb0145">[430db4333e]</a> Kevin Harwell -- chan_sip.c: wrong peer searched in sip_report_security_event</li>
-</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c7a1dca4baec5c2edec16470b377761066021650">[c7a1dca4ba]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d2e7419054cb4b837fa9113f0f309e19fc690ee">[2d2e741905]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
-</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25407">ASTERISK-25407</a>: Asterisk fails to log to multiple syslog destinations<br/>Reported by: Elazar Broad<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29694eb2aaa45c366fe28777722b0307f3742f95">[29694eb2aa]</a> Elazar Broad -- core/logging: Fix logging to more than one syslog channel</li>
-</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25394">ASTERISK-25394</a>: pbx: Incorrect device and presence state when changing hint details<br/>Reported by: Joshua Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51013b052d242717fce8b30bfba083721981b769">[51013b052d]</a> Joshua Colp -- pbx: Update device and presence state when changing a hint extension.</li>
-</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25312">ASTERISK-25312</a>: res_http_websocket: Terminate connection on fatal cases<br/>Reported by: Joshua Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9bd3c1435cb1e7b09851de2c7dcc5882bc2f9f4">[b9bd3c1435]</a> Joshua Colp -- res_http_websocket: Forcefully terminate on write errors.</li>
-</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c7a1dca4baec5c2edec16470b377761066021650">[c7a1dca4ba]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d2e7419054cb4b837fa9113f0f309e19fc690ee">[2d2e741905]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</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: Channels/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25427">ASTERISK-25427</a>: Callerid change does not always emit NewCallerid AMI event<br/>Reported by: Ivan Poddubny<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d2e1ecdca3291f0f82aac392bd68deec3c914e5">[8d2e1ecdca]</a> Ivan Poddubny -- channel.c: Fix NewCallerid AMI event not been sent on Caller ID change</li>
-</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25353">ASTERISK-25353</a>: [patch] Transcoding while different in Frame size = Frames lost<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c7f8c8c35db2fe1c4ce9f27c4a28649452dc5463">[c7f8c8c35d]</a> Alexander Traud -- translate: Fix transcoding while different in frame size.</li>
-</ul><br><h3>Improvement</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25310">ASTERISK-25310</a>: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ffa26a0c2e88ab294e6f0407faa71cc053a4e511">[ffa26a0c2e]</a> Guido Falsi -- Core/General: Add #ifdef needed on FreeBSD.</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=db337a8a50633167fced3c197372e1234a6ce67a">db337a8a50</a></td><td>Kevin Harwell</td><td>.version: Update for 11.20.0-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1db02c4e569935b233215ba10d81499e18880abc">1db02c4e56</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 11.20.0-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3cf0f293101cb8f044b3b1b57e3ecaaaf47365bf">3cf0f29310</a></td><td>Mark Michelson</td><td>scheduler: Use queue for allocating sched IDs.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=819760baece744384b7bba992738b002d50236e6">819760baec</a></td><td>David M. Lee</td><td>res_rtp_asterisk: Add more ICE debugging</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94b764c8f3e771c100154303c38949beaed9ea42">94b764c8f3</a></td><td>David M. Lee</td><td>Fix when remote candidates exceed PJ_ICE_MAX_CAND</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a364807f4b9016d0ccda9d67f1accacee7cbca5">6a364807f4</a></td><td>Richard Mudgett</td><td>app_queue.c: Extract some functions for simpler code.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a56da797d96bb797825b8c5029f597e1472050cc">a56da797d9</a></td><td>Richard Mudgett</td><td>app_queue.c: Fix error checking in QUEUE_MEMBER() read.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06b464ab1b7c42e3c1b5b05463d5bc73c112dd76">06b464ab1b</a></td><td>David M. Lee</td><td>Replace htobe64 with htonll</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5311d18101b9a1a20cb9b86a7102a1c7a0b1df28">5311d18101</a></td><td>Richard Mudgett</td><td>chan_sip.c: Move NULL check to where it will do some good.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75185c5d8f40e1bba48e15b566e4974967b261f6">75185c5d8f</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Fix off nominal ref leak and some minor tweaks.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b51b5efb619eb29c67ed5be0f4ab6d0793760d5">1b51b5efb6</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Tweak glue-&gt;update_peer() parameter nil value.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5cd1fa0df0b8ed9aa7736419f2b066c909bcadb">f5cd1fa0df</a></td><td>Richard Mudgett</td><td>chan_sip.c: Tweak glue-&gt;update_peer() parameter nil value.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f2089dce3dd972c147328cf8e7df5c756ee272b0">f2089dce3d</a></td><td>Mark Michelson</td><td>res_http_websocket: Properly encode 64 bit payload</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>ChangeLog                                |35699 -------------------------------
-asterisk-11.19.0-summary.html            |  139
-asterisk-11.19.0-summary.txt             |  445
-b/.version                               |    2
-b/addons/chan_ooh323.c                   |    1
-b/addons/ooh323c/src/ooq931.c            |    6
-b/apps/app_queue.c                       |  432
-b/apps/app_record.c                      |    3
-b/channels/chan_dahdi.c                  |   59
-b/channels/chan_sip.c                    |   77
-b/channels/sip/include/security_events.h |    3
-b/channels/sip/security_events.c         |    5
-b/codecs/codec_gsm.c                     |   29
-b/codecs/codec_ilbc.c                    |   28
-b/codecs/codec_lpc10.c                   |   41
-b/codecs/codec_speex.c                   |   61
-b/configure                              |   63
-b/configure.ac                           |    6
-b/contrib/scripts/install_prereq         |    2
-b/include/asterisk/autoconfig.h.in       |    3
-b/main/channel.c                         |   25
-b/main/logger.c                          |    7
-b/main/pbx.c                             |  129
-b/main/rtp_engine.c                      |   20
-b/main/sched.c                           |  171
-b/main/translate.c                       |   53
-b/main/utils.c                           |    4
-b/res/res_http_websocket.c               |   36
-b/res/res_rtp_asterisk.c                 |   52
-29 files changed, 931 insertions(+), 36670 deletions(-)</pre><br></html>
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-                                Release Summary
-
-                              asterisk-11.20.0-rc1
-
-                                Date: 2015-09-29
-
-                           <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-11.19.0.
-
-     ----------------------------------------------------------------------
-
-                                  Contributors
-
-                                 [Back to Top]
-
-   This table lists the people who have submitted code, those that have
-   tested patches, as well as those that reported issues on the issue tracker
-   that were resolved in this release. For coders, the number is how many of
-   their patches (of any size) were committed into this release. For testers,
-   the number is the number of times their name was listed as assisting with
-   testing a patch. Finally, for reporters, the number is the number of
-   issues that they reported that were affected by commits that went into
-   this release.
-
-   Coders                   Testers                  Reporters                
-   9 Richard Mudgett        1 Elazar Broad           3 Joshua Colp            
-   4 Joshua Colp                                     2 Stefan EngstrAP:m      
-   3 David M. Lee                                    2 Richard Mudgett        
-   2 Kevin Harwell                                   2 John Hardin            
-   2 Mark Michelson                                  2 Alexandr Dranchuk      
-   2 Kevin Harwell                                   2 Kevin Harwell          
-   2 Alexander Anikin                                1 Guido Falsi            
-   1 Walter Doekes                                   1 Walter Doekes          
-   1 Ivan Poddubny                                   1 Kevin Harwell          
-   1 Elazar Broad                                    1 Ivan Poddubny          
-   1 Mark Duncan                                     1 Ivan Poddubny          
-   1 Scott Griepentrog                               1 Elazar Broad           
-   1 Guido Falsi                                     1 Lorne Gaetz            
-   1 Alexander Traud                                 1 Scott Griepentrog      
-                                                     1 Elazar Broad           
-                                                     1 Guido Falsi            
-                                                     1 Lorne Gaetz            
-                                                     1 Alexandr Dranchuk      
-                                                     1 Alexander Traud        
-
-     ----------------------------------------------------------------------
-
-                                 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: Addons/chan_ooh323
-
-   ASTERISK-25227: No audio at in-band announcements in ooh323 channel
-   Reported by: Alexandr Dranchuk
-     * [198a1cab8e] Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE
-       with inband info available
-   ASTERISK-25299: RTP port leaks with incoming OOH323 calls
-   Reported by: Alexandr Dranchuk
-     * [07b25a2312] Alexander Anikin -- chan_ooh323: call
-       ast_rtp_instance_stop on ooh323_destroy
-
-    Category: Applications/app_queue
-
-   ASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and
-   using PauseQueueMember
-   Reported by: Lorne Gaetz
-     * [43150cc58d] Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER
-       'paused' and 'ringinuse'.
-
-    Category: Applications/app_record
-
-   ASTERISK-25410: app_record: RECORDED_FILE variable not being populated
-   Reported by: Kevin Harwell
-     * [455a31476b] Kevin Harwell -- app_record: RECORDED_FILE variable not
-       being populated
-
-    Category: Channels/chan_dahdi
-
-   ASTERISK-25315: DAHDI channels send shortened duration DTMF tones.
-   Reported by: Richard Mudgett
-     * [c777c9565d] Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write
-       buffer after starting DTMF.
-     * [f43ea74e9e] Richard Mudgett -- chan_dahdi.c: Lock private struct for
-       ast_write().
-
-    Category: Channels/chan_sip/General
-
-   ASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on
-   call pickup
-   Reported by: Joshua Colp
-     * [59636e82b2] Joshua Colp -- chan_sip: Allow call pickup to set the
-       hangup cause.
-
-    Category: Channels/chan_sip/Interoperability
-
-   ASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP
-   Reported by: Walter Doekes
-     * [2fcf45e9cf] Walter Doekes -- chan_sip: Fix From header truncation for
-       extremely long CALLERID(name).
-
-    Category: Channels/chan_sip/Security Framework
-
-   ASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong
-   or non existent peer on invite
-   Reported by: Kevin Harwell
-     * [430db4333e] Kevin Harwell -- chan_sip.c: wrong peer searched in
-       sip_report_security_event
-
-    Category: Core/BuildSystem
-
-   ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
-   - add ECDH support and fallback to prime256v1
-   Reported by: Stefan EngstrAP:m
-     * [c7a1dca4ba] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
-       when enabling PFS.
-     * [2d2e741905] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
-
-    Category: Core/Logging
-
-   ASTERISK-25407: Asterisk fails to log to multiple syslog destinations
-   Reported by: Elazar Broad
-     * [29694eb2aa] Elazar Broad -- core/logging: Fix logging to more than
-       one syslog channel
-
-    Category: Core/PBX
-
-   ASTERISK-25394: pbx: Incorrect device and presence state when changing
-   hint details
-   Reported by: Joshua Colp
-     * [51013b052d] Joshua Colp -- pbx: Update device and presence state when
-       changing a hint extension.
-
-    Category: Resources/res_http_websocket
-
-   ASTERISK-25312: res_http_websocket: Terminate connection on fatal cases
-   Reported by: Joshua Colp
-     * [b9bd3c1435] Joshua Colp -- res_http_websocket: Forcefully terminate
-       on write errors.
-
-    Category: Resources/res_rtp_asterisk
-
-   ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
-   - add ECDH support and fallback to prime256v1
-   Reported by: Stefan EngstrAP:m
-     * [c7a1dca4ba] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
-       when enabling PFS.
-     * [2d2e741905] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
-
-     ----------------------------------------------------------------------
-
-                                  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: Channels/General
-
-   ASTERISK-25427: Callerid change does not always emit NewCallerid AMI event
-   Reported by: Ivan Poddubny
-     * [8d2e1ecdca] Ivan Poddubny -- channel.c: Fix NewCallerid AMI event not
-       been sent on Caller ID change
-
-    Category: Codecs/General
-
-   ASTERISK-25353: [patch] Transcoding while different in Frame size = Frames
-   lost
-   Reported by: Alexander Traud
-     * [c7f8c8c35d] Alexander Traud -- translate: Fix transcoding while
-       different in frame size.
-
-  Improvement
-
-    Category: Core/General
-
-   ASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to
-   PTHREAD_EXPLICIT_SCHED
-   Reported by: Guido Falsi
-     * [ffa26a0c2e] Guido Falsi -- Core/General: Add #ifdef needed on
-       FreeBSD.
-
-     ----------------------------------------------------------------------
-
-                      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                                 |
-   |------------+-----------------+-----------------------------------------|
-   | db337a8a50 | Kevin Harwell   | .version: Update for 11.20.0-rc1        |
-   |------------+-----------------+-----------------------------------------|
-   | 1db02c4e56 | Kevin Harwell   | .lastclean: Update for 11.20.0-rc1      |
-   |------------+-----------------+-----------------------------------------|
-   | 3cf0f29310 | Mark Michelson  | scheduler: Use queue for allocating     |
-   |            |                 | sched IDs.                              |
-   |------------+-----------------+-----------------------------------------|
-   | 819760baec | David M. Lee    | res_rtp_asterisk: Add more ICE          |
-   |            |                 | debugging                               |
-   |------------+-----------------+-----------------------------------------|
-   | 94b764c8f3 | David M. Lee    | Fix when remote candidates exceed       |
-   |            |                 | PJ_ICE_MAX_CAND                         |
-   |------------+-----------------+-----------------------------------------|
-   | 6a364807f4 | Richard Mudgett | app_queue.c: Extract some functions for |
-   |            |                 | simpler code.                           |
-   |------------+-----------------+-----------------------------------------|
-   | a56da797d9 | Richard Mudgett | app_queue.c: Fix error checking in      |
-   |            |                 | QUEUE_MEMBER() read.                    |
-   |------------+-----------------+-----------------------------------------|
-   | 06b464ab1b | David M. Lee    | Replace htobe64 with htonll             |
-   |------------+-----------------+-----------------------------------------|
-   | 5311d18101 | Richard Mudgett | chan_sip.c: Move NULL check to where it |
-   |            |                 | will do some good.                      |
-   |------------+-----------------+-----------------------------------------|
-   | 75185c5d8f | Richard Mudgett | rtp_engine.c: Fix off nominal ref leak  |
-   |            |                 | and some minor tweaks.                  |
-   |------------+-----------------+-----------------------------------------|
-   | 1b51b5efb6 | Richard Mudgett | rtp_engine.c: Tweak glue->update_peer() |
-   |            |                 | parameter nil value.                    |
-   |------------+-----------------+-----------------------------------------|
-   | f5cd1fa0df | Richard Mudgett | chan_sip.c: Tweak glue->update_peer()   |
-   |            |                 | parameter nil value.                    |
-   |------------+-----------------+-----------------------------------------|
-   | f2089dce3d | Mark Michelson  | res_http_websocket: Properly encode 64  |
-   |            |                 | bit payload                             |
-   +------------------------------------------------------------------------+
-
-     ----------------------------------------------------------------------
-
-                                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.
-
- ChangeLog                                |35699 -------------------------------
- asterisk-11.19.0-summary.html            |  139
- asterisk-11.19.0-summary.txt             |  445
- b/.version                               |    2
- b/addons/chan_ooh323.c                   |    1
- b/addons/ooh323c/src/ooq931.c            |    6
- b/apps/app_queue.c                       |  432
- b/apps/app_record.c                      |    3
- b/channels/chan_dahdi.c                  |   59
- b/channels/chan_sip.c                    |   77
- b/channels/sip/include/security_events.h |    3
- b/channels/sip/security_events.c         |    5
- b/codecs/codec_gsm.c                     |   29
- b/codecs/codec_ilbc.c                    |   28
- b/codecs/codec_lpc10.c                   |   41
- b/codecs/codec_speex.c                   |   61
- b/configure                              |   63
- b/configure.ac                           |    6
- b/contrib/scripts/install_prereq         |    2
- b/include/asterisk/autoconfig.h.in       |    3
- b/main/channel.c                         |   25
- b/main/logger.c                          |    7
- b/main/pbx.c                             |  129
- b/main/rtp_engine.c                      |   20
- b/main/sched.c                           |  171
- b/main/translate.c                       |   53
- b/main/utils.c                           |    4
- b/res/res_http_websocket.c               |   36
- b/res/res_rtp_asterisk.c                 |   52
- 29 files changed, 931 insertions(+), 36670 deletions(-)