Update for 17.2.0

This commit is contained in:
Asterisk Development Team 2020-02-04 10:08:19 -05:00
parent 098c164ff7
commit 758a5207e4
6 changed files with 826 additions and 112 deletions

View File

@ -1 +1 @@
17.2.0-rc2
17.2.0

View File

@ -1,3 +1,7 @@
2020-02-04 15:08 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 17.2.0 Released.
2020-01-30 16:38 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 17.2.0-rc2 Released.

View File

@ -1,15 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-17.2.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.2.0-rc2</h3><h3 align="center">Date: 2020-01-30</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="#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-17.2.0-rc1.</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%">2 Kevin Harwell <kharwell@digium.com><br/>1 Joshua C. Colp <jcolp@sangoma.com><br/></td><td width="33%"><td width="33%">2 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Francois Blackburn <fblackburn@wazo.io><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: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28679">ASTERISK-28679</a>: stasis application is destroyed after its creation<br/>Reported by: Francois Blackburn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=19b0efce0d5c86b833904689d51cde772a1b4c4d">[19b0efce0d]</a> Kevin Harwell -- res_stasis: trigger cleanup after update</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28714">ASTERISK-28714</a>: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b4d900e4317d2fc2d6270848ea3b189cce17d96">[5b4d900e43]</a> Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78a907e23d9a40381e00ebf31e5156fcb091ac81">[78a907e23d]</a> Kevin Harwell -- stasis/app: don't lock an app before a call to send</li>
</ul><br><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>0 files changed</pre><br></html>

View File

@ -1,96 +0,0 @@
Release Summary
asterisk-17.2.0-rc2
Date: 2020-01-30
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. 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-17.2.0-rc1.
----------------------------------------------------------------------
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
2 Kevin Harwell 2 Ross Beer
1 Joshua C. Colp 1 Francois Blackburn
----------------------------------------------------------------------
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: Resources/res_ari
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
* [19b0efce0d] Kevin Harwell -- res_stasis: trigger cleanup after update
Category: Resources/res_pjsip_pubsub
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate
(ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
* [5b4d900e43] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence
data ref when recreating.
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
* [78a907e23d] Kevin Harwell -- stasis/app: don't lock an app before a
call to send
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
0 files changed

View File

@ -0,0 +1,201 @@
<!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-17.2.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.2.0</h3><h3 align="center">Date: 2020-02-04</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-17.1.0.</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%">20 Sean Bright <sean.bright@gmail.com><br/>7 George Joseph <gjoseph@digium.com><br/>6 Richard Mudgett <rmudgett@digium.com><br/>5 Joshua C. Colp <jcolp@sangoma.com><br/>4 Kevin Harwell <kharwell@digium.com><br/>3 Asterisk Development Team <asteriskteam@digium.com><br/>3 Jaco Kroon <jaco@uls.co.za><br/>2 Pascal Cadotte Michaud <pcm@wazo.io><br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Joshua Colp <jcolp@sangoma.com><br/>1 Jean Aunis <jean.aunis@prescom.fr><br/>1 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Corey Farrell <git@cfware.com><br/>1 snuffy <snuffy22@gmail.com><br/></td><td width="33%"><td width="33%">3 Sean Bright <sean.bright@gmail.com><br/>3 Ross Beer <ross.beer@voicehost.co.uk><br/>3 cmaj <chris@penguinpbx.com><br/>2 Pascal Cadotte Michaud <pascal.cadotte@gmail.com><br/>2 Joshua C. Colp <jcolp@digium.com><br/>1 Robert Sutton <rsutton@noojee.com.au><br/>1 Kevin Flyn<br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Maciej Michno <maciej.michno@xtb.com><br/>1 AvayaXAsterisk<br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 George Joseph <gjoseph@digium.com><br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 candrews <candrews@integralblue.com><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Frank Matano <ftalarico99@gmail.com><br/>1 Cédric Bassaget<br/>1 Kevin Harwell <kharwell@digium.com><br/>1 Dan Jenkins <dan@nimbleape.com><br/>1 kevin@phoneburner.com<br/>1 Maciej Michno<br/>1 Dirk Wendland <dirk@starface.de><br/>1 Jim Van Meggelen <jim.vanmeggelen@clearlycore.com><br/>1 Ted G<br/>1 Stas Kobzar<br/>1 Jean-Denis Girard<br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Cedric BASSAGET <cedric@oceanet.com><br/>1 Niksa Baldun <niksa.baldun@gmail.com><br/>1 Ted G <tgwaste@gmail.com><br/>1 Corey Farrell <git@cfware.com><br/>1 Richard Kenner<br/>1 Frank Matano<br/>1 Joeran Vinzens <vinzens@sipgate.de><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Kevin Flyn <kevflynn69@gmail.com><br/>1 David M. Lee <dlee@digium.com><br/>1 Dirk Wendland<br/>1 Bryan Nelson <bnelson@fluentstream.com><br/>1 Richard Kenner <kenner@gnat.com><br/>1 Ross Beer<br/>1 Francois Blackburn <fblackburn@wazo.io><br/>1 Joeran Vinzens<br/>1 Jonathan Harris <lardconcepts@gmail.com><br/>1 Jonathan Harris<br/>1 Dan Jenkins<br/>1 AvayaXAsterisk <joh.zuerner@yahoo.de><br/>1 Sean Bright<br/>1 Jean-Denis Girard <jd.girard@sysnux.pf><br/>1 nappsoft <infos@nappsoft.ch><br/>1 Joshua C. Colp<br/>1 Niksa Baldun<br/>1 Mitch Claborn<br/>1 Robin Leffmann <robin@stolendata.net><br/>1 Jim Van Meggelen<br/>1 David Lee<br/>1 Robert Sutton<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>New Feature</h3><h4>Category: Functions/func_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17491">ASTERISK-17491</a>: CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything<br/>Reported by: candrews<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a8100257821d890e1384ce4d8a41cf84782ff161">[a810025782]</a> Sean Bright -- func_curl: Add 'followlocation' option to CURLOPT()</li>
</ul><br><h4>Category: Resources/res_pjsip_endpoint_identifier_ip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28639">ASTERISK-28639</a>: res_pjsip_endpoint_identifier_ip: Add ability to match on source port<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eac6eb663d3ad4803be24c48ecbb9f18fba145c3">[eac6eb663d]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add port matching support</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_chanisavail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28636">ASTERISK-28636</a>: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0e31cc14e524f720aa80beaac9fcb9fff5e4b24">[a0e31cc14e]</a> Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28604">ASTERISK-28604</a>: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a80c36d624d375b3d28f845238e529f61c59fb1">[5a80c36d62]</a> Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28349">ASTERISK-28349</a>: Pause reason not reported in QueueMember AMI event<br/>Reported by: Niksa Baldun<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62e5fa400e12be2b23cea4799491309d91906b78">[62e5fa400e]</a> Sean Bright -- app_queue: Deprecate the QueueMemberPause.Reason field</li>
</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28682">ASTERISK-28682</a>: app_record: Lack of `beep` audio file causes application to return error and hangup<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1675ab3ce25ca60549bac510b66c494db692a638">[1675ab3ce2]</a> Corey Farrell -- app_record: Do not hang up if beep audio is missing</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23739">ASTERISK-23739</a>: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used<br/>Reported by: Stas Kobzar<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1cd27ba1f7a679639c06a23cbec73d5a2714329">[d1cd27ba1f]</a> Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27622">ASTERISK-27622</a>: empty voicemail.conf required for ARA (realtime) voicemail to leave message<br/>Reported by: Jim Van Meggelen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b60dfac98ec6fa04e57bccafe0b7fe76778f8546">[b60dfac98e]</a> Sean Bright -- app_voicemail: Set globals to default values when voicemail.conf missing</li>
</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23739">ASTERISK-23739</a>: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used<br/>Reported by: Stas Kobzar<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1cd27ba1f7a679639c06a23cbec73d5a2714329">[d1cd27ba1f]</a> Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail</li>
</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28677">ASTERISK-28677</a>: CDR billsec is always 0 for transferred calls<br/>Reported by: Maciej Michno<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f34a4d5a775e2dd3feba11102c9b23fbff12eb72">[f34a4d5a77]</a> George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28636">ASTERISK-28636</a>: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0e31cc14e524f720aa80beaac9fcb9fff5e4b24">[a0e31cc14e]</a> Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28702">ASTERISK-28702</a>: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40<br/>Reported by: Andrew Siplas<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1c5116f8421a8d303c870068062984e7f0053334">[1c5116f842]</a> Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28492">ASTERISK-28492</a>: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group<br/>Reported by: Jean-Denis Girard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0d1197ab384d50bf96a38f53b6cfa593f6771a2">[a0d1197ab3]</a> Sean Bright -- res_pjsip_config_wizard: Fix change detection for wizard settings</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28502">ASTERISK-28502</a>: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd14d7302f0c1fa748bdaa0d878f9377c0e60735">[fd14d7302f]</a> George Joseph -- res_pjsip_nat: Restore original contact for REGISTER responses</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28647">ASTERISK-28647</a>: chan_sip: RTP frames not transmitted after emitting a COLP<br/>Reported by: Jean Aunis - Prescom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=811c7bdabe90e52cef9767a70def012200d5a4de">[811c7bdabe]</a> Jean Aunis -- chan_sip: voice frames are no longer transmitted after emitting a COLP</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28651">ASTERISK-28651</a>: chan_sip logs errors on tx to non-existent TCP connections<br/>Reported by: Jaco Kroon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32ce8fa0463ca35b4afc5d38c6fd11dddf58c3bd">[32ce8fa046]</a> Jaco Kroon -- chan_sip: in case of tcp/tls, be less annoying about tx errors.</li>
</ul><br><h4>Category: Channels/chan_sip/Messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28693">ASTERISK-28693</a>: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan<br/>Reported by: Frank Matano<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c137ebbd46ae4eb957d591b87851f7d6d6c1d17">[7c137ebbd4]</a> Sean Bright -- chan_sip.c: Stop handling continuation lines after reading headers</li>
</ul><br><h4>Category: Channels/chan_sip/Transfers</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28677">ASTERISK-28677</a>: CDR billsec is always 0 for transferred calls<br/>Reported by: Maciej Michno<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f34a4d5a775e2dd3feba11102c9b23fbff12eb72">[f34a4d5a77]</a> George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge</li>
</ul><br><h4>Category: Codecs/codec_silk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28706">ASTERISK-28706</a>: silk 24hHz doesn't show up in 'core show translation' output<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=687890f73400a0a8d847a86f029d982bb97273cf">[687890f734]</a> Sean Bright -- translate.c: Fix silk 24kHz truncation in 'core show translation'</li>
</ul><br><h4>Category: Configs/Basic-PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28667">ASTERISK-28667</a>: Asterisk ignores parsing of config files if a Byte order mark is present<br/>Reported by: Robin Leffmann<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c8989f205b24e13c038046d90dff7321b42748b">[8c8989f205]</a> Sean Bright -- config.c: Skip UTF-8 BOMs if present when reading config files</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27243">ASTERISK-27243</a>: contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax<br/>Reported by: Richard Kenner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c4cd2fa7b9e6bd8f94d86c9afa0ba7eef8c2857a">[c4cd2fa7b9]</a> snuffy -- contrib/valgrind: Fix use of frame-level suppression</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28664">ASTERISK-28664</a>: "trustrpid" is misspelled in sip_to_pjsip.py<br/>Reported by: Pascal Cadotte Michaud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=69c9b74836c038761d2d764678636a8212217ef7">[69c9b74836]</a> Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid typo</li>
</ul><br><h4>Category: Core/Streams</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28625">ASTERISK-28625</a>: Playback of local files impacted by large media cache<br/>Reported by: Kevin Reeves<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45f7f9bb85fede58fcf594d6355a43c2fffe7201">[45f7f9bb85]</a> Kevin Reeves -- main/file.c: Limit media cache usage to remote files.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24484">ASTERISK-24484</a>: Update documentation for statsd module - usage requirements unclear<br/>Reported by: Dan Jenkins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=deed0f5706749e882390fdd5af3d11b22dc860d6">[deed0f5706]</a> Sean Bright -- res_statsd: Document that res_statsd does nothing on its own</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25429">ASTERISK-25429</a>: res_pjsip_endpoint_identifier_ip: Document support for hostnames<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28275609b3f01266b45da90f3f12ed2ac55cb364">[28275609b3]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28507">ASTERISK-28507</a>: Wiki docs missing for MessageWaiting<br/>Reported by: David M. Lee<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40bfd895a988e18fa9e9f5b7907a9875caf88fa8">[40bfd895a9]</a> George Joseph -- CI: Update buildAsterisk.sh to do a "make full"</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28626">ASTERISK-28626</a>: Missing arguments in PJSIP_CONTACT function documentation<br/>Reported by: Pascal Cadotte Michaud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39aff53cff244d0272db9f91ef3453dc7bdf0556">[39aff53cff]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</li>
</ul><br><h4>Category: Functions/func_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28497">ASTERISK-28497</a>: func_odbc: truncating Unicode string on readsql<br/>Reported by: Boris P. Korzun<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d2eced3d3b5ed84530292e284227f1604246b24">[6d2eced3d3]</a> Boris P. Korzun -- func_odbc: acf_odbc_read() and cli_odbc_read() unicode support</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28609">ASTERISK-28609</a>: Memory Leak in res_rtp_asterisk.c<br/>Reported by: Ted G<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86822722b625326a33fe22f6a9c561bda62b14e5">[86822722b6]</a> George Joseph -- res_rtp_asterisk: Add frame list cleanups to ast_rtp_read</li>
</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28695">ASTERISK-28695</a>: core: minmemfree watermark uses free RAM, not available RAM<br/>Reported by: Kevin Flyn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ff062f99435949bfe627eefbf94549a01d3e455">[9ff062f994]</a> Sean Bright -- pbx.c: Include filesystem cache in free memory calculation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28605">ASTERISK-28605</a>: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X<br/>Reported by: Dirk Wendland<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=015cdc9f64ffa8545376b96ced8704ebfc585ad3">[015cdc9f64]</a> George Joseph -- sig_pri: Fix deadlock caused by sig_pri_queue_hangup</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28679">ASTERISK-28679</a>: stasis application is destroyed after its creation<br/>Reported by: Francois Blackburn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=19b0efce0d5c86b833904689d51cde772a1b4c4d">[19b0efce0d]</a> Kevin Harwell -- res_stasis: trigger cleanup after update</li>
</ul><br><h4>Category: Resources/res_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28660">ASTERISK-28660</a>: res_fax: wrap Asterisk initiated negotiation with config option<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68724425e435f911c8549db6ac38659486e7b755">[68724425e4]</a> Kevin Harwell -- res_fax: wrap v21 detected Asterisk initiated negotiation with config option</li>
</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28562">ASTERISK-28562</a>: SIP WSS message not processed until next frame arrives<br/>Reported by: Robert Sutton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f3443562901524a04c6dff5010d234b2468df15e">[f344356290]</a> Sean Bright -- websocket: Consider pending SSL data when waiting for socket input</li>
</ul><br><h4>Category: Resources/res_pjsip_endpoint_identifier_ip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25429">ASTERISK-25429</a>: res_pjsip_endpoint_identifier_ip: Document support for hostnames<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28275609b3f01266b45da90f3f12ed2ac55cb364">[28275609b3]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames</li>
</ul><br><h4>Category: Resources/res_pjsip_notify</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27775">ASTERISK-27775</a>: res_pjsip_notify: Multiple Event headers can be present instead of just one<br/>Reported by: AvayaXAsterisk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6898c8e0daedc17b768de6c9ae9ed1ce74d5cd41">[6898c8e0da]</a> Sean Bright -- res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28714">ASTERISK-28714</a>: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b4d900e4317d2fc2d6270848ea3b189cce17d96">[5b4d900e43]</a> Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27759">ASTERISK-27759</a>: res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number<br/>Reported by: Bryan Nelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e17ba921f3b3c2b677b5229fdf0ba78a22251506">[e17ba921f3]</a> Joshua C. Colp -- res_pjsip_pubsub: Add ability to persist generator state information.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c9147a759e85ea2826e38644df290f422ce65603">[c9147a759e]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c9147a759e85ea2826e38644df290f422ce65603">[c9147a759e]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
</ul><br><h4>Category: Resources/res_realtime</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21794">ASTERISK-21794</a>: CLI command 'realtime update2' syntax failure when using according to usage help<br/>Reported by: Cedric BASSAGET<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3779e76b6892c4b33746ec3835c8821146cd1e0d">[3779e76b68]</a> Sean Bright -- res_realtime: Fix 'realtime update2' argument handling</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78a907e23d9a40381e00ebf31e5156fcb091ac81">[78a907e23d]</a> Kevin Harwell -- stasis/app: don't lock an app before a call to send</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28633">ASTERISK-28633</a>: stasis bridge topic leak<br/>Reported by: Joeran Vinzens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0449939ec6a5950556e3f6b8e14ef4a370c2849">[e0449939ec]</a> George Joseph -- stasis.c: Use correct topic name in stasis_topic_pool_delete_topic</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24484">ASTERISK-24484</a>: Update documentation for statsd module - usage requirements unclear<br/>Reported by: Dan Jenkins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=deed0f5706749e882390fdd5af3d11b22dc860d6">[deed0f5706]</a> Sean Bright -- res_statsd: Document that res_statsd does nothing on its own</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28658">ASTERISK-28658</a>: app_confbridge: Add support for setting maximum sample rate<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41fdcf7a65f13008c05b3c72a2674031aee535a9">[41fdcf7a65]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28658">ASTERISK-28658</a>: app_confbridge: Add support for setting maximum sample rate<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41fdcf7a65f13008c05b3c72a2674031aee535a9">[41fdcf7a65]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28638">ASTERISK-28638</a>: Simplify dialplan for Dial, Page, and ChanIsAvail<br/>Reported by: cmaj<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62a8750d2dbe62123bd326e1dc5dbaa8812a2e5e">[62a8750d2d]</a> Richard Mudgett -- app_chanisavail.c: Simplify dialplan using ChanIsAvail.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3d0b0c484caded330124c6fcc601422b3421910">[b3d0b0c484]</a> Richard Mudgett -- app_dial.c: Simplify dialplan using Dial.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7c6a7df9f64f800daee96c2181e049cbf13dd1f">[f7c6a7df9f]</a> Richard Mudgett -- app_page.c: Simplify dialplan using Page.</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28710">ASTERISK-28710</a>: Should be able to disable the /httpstatus URI in the built-in HTTP server<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e79bf44fc6741df4d325e9659b0763d0e09ef02">[1e79bf44fc]</a> Sean Bright -- http: Add ability to disable /httpstatus URI</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28673">ASTERISK-28673</a>: GET FULL VARIABLE documentation clarification<br/>Reported by: Jonathan Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7038c8676e3dd794786779bb8218fc026825277e">[7038c8676e]</a> Sean Bright -- res_agi: Improve GET FULL VARIABLE documentation</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=098c164ff7b798cf1dcd42aecedb0d6d573e8250">098c164ff7</a></td><td>Joshua Colp</td><td>REVERT: Add option to suppress the Message channel AMI and ARI events</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bdcf8e71655cd2fd79264e212cb50494c70c1e05">bdcf8e7165</a></td><td>George Joseph</td><td>message.c: Add option to suppress the Message channel AMI and ARI events</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b027e249029fcce322dbda430439ef66bafa1d6">8b027e2490</a></td><td>Asterisk Development Team</td><td>Update for 17.2.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4579ddd8466d4878c628650b56d80b7226967703">4579ddd846</a></td><td>Asterisk Development Team</td><td>Update for 17.2.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0f44b10522b92e0b684a9cb1f35325047b9f89f">a0f44b1052</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.2.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d33cd39260325f550606682a36e4ce8ba5dbca66">d33cd39260</a></td><td>Sean Bright</td><td>func_odbc.conf.sample: Add example lookup</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b327631903d308c661285edf4a2ade2d8f47f84">7b32763190</a></td><td>Rodrigo Ramírez Norambuena</td><td>queue_log: Add alembic script for generate db table for queue_log</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb6f106d7604ea408b92d3882c367ff46f4af645">cb6f106d76</a></td><td>Sean Bright</td><td>app_voicemail, say: Fix various leading whitespace problems</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a9e5febd5ab753677ad8f7b2dd223762be609f5">0a9e5febd5</a></td><td>Jaco Kroon</td><td>netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21d8d2426d2262f985702efbb5addb22a31141a8">21d8d2426d</a></td><td>Kevin Harwell</td><td>app_agent_pool: Update XML docs for AgentLogin</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=919da21690667c597bd6ff9b82f1da6aacfeeaa3">919da21690</a></td><td>Richard Mudgett</td><td>features.c: Make Bridge application tolerate unspecified channel.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a82fc356065ba933dd4c9e13a5455bff5e24ccb9">a82fc35606</a></td><td>Richard Mudgett</td><td>app_chanspy.c: Reduce log message level from notice to verbose.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4dfa19eb273a571b8a0b05c6eaaaf5caadda9028">4dfa19eb27</a></td><td>Richard Mudgett</td><td>app_softhangup.c: Reduce unnecessary warning to verbose message.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=843d601e2cf8d1a081fc1e229d3dbb67dd67e80f">843d601e2c</a></td><td>Sean Bright</td><td>db: Initialize condition primitive before use</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6041fc8b9cf032e97c72c8e4c9656a1952383adf">6041fc8b9c</a></td><td>Jaco Kroon</td><td>ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging.</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-17.1.0-summary.html | 372 ---
asterisk-17.1.0-summary.txt | 962 ----------
b/.version | 2
b/CHANGES | 55
b/ChangeLog | 834 ++++++++
b/apps/app_agent_pool.c | 4
b/apps/app_chanisavail.c | 140 -
b/apps/app_chanspy.c | 3
b/apps/app_confbridge.c | 2
b/apps/app_dial.c | 51
b/apps/app_page.c | 30
b/apps/app_queue.c | 2
b/apps/app_record.c | 3
b/apps/app_softhangup.c | 2
b/apps/app_voicemail.c | 407 ++--
b/apps/confbridge/conf_config_parser.c | 17
b/apps/confbridge/include/confbridge.h | 1
b/asterisk-17.2.0-rc2-summary.html | 15
b/asterisk-17.2.0-rc2-summary.txt | 96
b/bridges/bridge_softmix.c | 18
b/channels/chan_dahdi.c | 2
b/channels/chan_sip.c | 30
b/channels/sig_pri.c | 23
b/configs/samples/confbridge.conf.sample | 4
b/configs/samples/func_odbc.conf.sample | 8
b/configs/samples/http.conf.sample | 10
b/configs/samples/pjsip.conf.sample | 1
b/contrib/ast-db-manage/README.md | 1
b/contrib/ast-db-manage/queue_log.ini.sample | 58
b/contrib/ast-db-manage/queue_log/env.py | 1
b/contrib/ast-db-manage/queue_log/script.py.mako | 24
b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py | 38
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/contrib/valgrind.supp | 14
b/doc/CHANGES-staging/res_fax_negotiate_both | 7
b/doc/appdocsxml.dtd | 2
b/funcs/func_curl.c | 11
b/funcs/func_odbc.c | 22
b/funcs/func_pjsip_contact.c | 6
b/include/asterisk/acl.h | 37
b/include/asterisk/bridge.h | 21
b/include/asterisk/http_websocket.h | 14
b/include/asterisk/iostream.h | 14
b/include/asterisk/netsock2.h | 42
b/include/asterisk/res_fax.h | 3
b/include/asterisk/res_pjsip_pubsub.h | 23
b/include/asterisk/stasis.h | 3
b/main/acl.c | 74
b/main/bridge.c | 7
b/main/cdr.c | 15
b/main/config.c | 12
b/main/db.c | 3
b/main/features.c | 28
b/main/file.c | 7
b/main/http.c | 56
b/main/iostream.c | 14
b/main/pbx.c | 12
b/main/say.c | 956 ++++-----
b/main/stasis.c | 17
b/main/translate.c | 8
b/res/res_agi.c | 20
b/res/res_fax.c | 26
b/res/res_http_websocket.c | 11
b/res/res_pjsip/pjsip_message_filter.c | 40
b/res/res_pjsip_config_wizard.c | 7
b/res/res_pjsip_dialog_info_body_generator.c | 80
b/res/res_pjsip_endpoint_identifier_ip.c | 86
b/res/res_pjsip_nat.c | 84
b/res/res_pjsip_notify.c | 22
b/res/res_pjsip_pubsub.c | 87
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_realtime.c | 56
b/res/res_stasis.c | 8
b/res/res_statsd.c | 35
74 files changed, 2848 insertions(+), 2362 deletions(-)</pre><br></html>

620
asterisk-17.2.0-summary.txt Normal file
View File

@ -0,0 +1,620 @@
Release Summary
asterisk-17.2.0
Date: 2020-02-04
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-17.1.0.
----------------------------------------------------------------------
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
20 Sean Bright 3 Sean Bright
7 George Joseph 3 Ross Beer
6 Richard Mudgett 3 cmaj
5 Joshua C. Colp 2 Pascal Cadotte Michaud
4 Kevin Harwell 2 Joshua C. Colp
3 Asterisk Development Team 1 Robert Sutton
3 Jaco Kroon 1 Kevin Flyn
2 Pascal Cadotte Michaud 1 Kevin Reeves
1 Kevin Reeves 1 Maciej Michno
1 Frederic LE FOLL 1 AvayaXAsterisk
1 Joshua Colp 1 Jean Aunis - Prescom
1 Jean Aunis 1 George Joseph
1 Rodrigo RamÃrez Norambuena 1 Jaco Kroon
1 Boris P. Korzun 1 candrews
1 Andrew Siplas 1 Andrew Siplas
1 Corey Farrell 1 Frank Matano
1 snuffy 1 Cédric Bassaget
1 Kevin Harwell
1 Dan Jenkins
1 kevin@phoneburner.com
1 Maciej Michno
1 Dirk Wendland
1 Jim Van Meggelen
1 Ted G
1 Stas Kobzar
1 Jean-Denis Girard
1 Stas Kobzar
1 Boris P. Korzun
1 Cedric BASSAGET
1 Niksa Baldun
1 Ted G
1 Corey Farrell
1 Richard Kenner
1 Frank Matano
1 Joeran Vinzens
1 Frederic LE FOLL
1 Kevin Flyn
1 David M. Lee
1 Dirk Wendland
1 Bryan Nelson
1 Richard Kenner
1 Ross Beer
1 Francois Blackburn
1 Joeran Vinzens
1 Jonathan Harris
1 Jonathan Harris
1 Dan Jenkins
1 AvayaXAsterisk
1 Sean Bright
1 Jean-Denis Girard
1 nappsoft
1 Joshua C. Colp
1 Niksa Baldun
1 Mitch Claborn
1 Robin Leffmann
1 Jim Van Meggelen
1 David Lee
1 Robert Sutton
----------------------------------------------------------------------
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.
New Feature
Category: Functions/func_curl
ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs"
doesn't do anything
Reported by: candrews
* [a810025782] Sean Bright -- func_curl: Add 'followlocation' option to
CURLOPT()
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on
source port
Reported by: Sean Bright
* [eac6eb663d] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add
port matching support
Bug
Category: Applications/app_chanisavail
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to
deactivate CDR.
Reported by: Frederic LE FOLL
* [a0e31cc14e] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail
sometimes fails to deactivate CDR.
Category: Applications/app_meetme
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
17.0.0
Reported by: George Joseph
* [5a80c36d62] Joshua C. Colp -- configure: Add check for MySQL client
bool and my_bool type usage.
Category: Applications/app_queue
ASTERISK-28349: Pause reason not reported in QueueMember AMI event
Reported by: Niksa Baldun
* [62e5fa400e] Sean Bright -- app_queue: Deprecate the
QueueMemberPause.Reason field
Category: Applications/app_record
ASTERISK-28682: app_record: Lack of `beep` audio file causes application
to return error and hangup
Reported by: Corey Farrell
* [1675ab3ce2] Corey Farrell -- app_record: Do not hang up if beep audio
is missing
Category: Applications/app_voicemail
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage
enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
* [d1cd27ba1f] Sean Bright -- app_voicemail: Prevent crash when saving
message with realtime voicemail
ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail
to leave message
Reported by: Jim Van Meggelen
* [b60dfac98e] Sean Bright -- app_voicemail: Set globals to default
values when voicemail.conf missing
Category: Applications/app_voicemail/ODBC
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage
enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
* [d1cd27ba1f] Sean Bright -- app_voicemail: Prevent crash when saving
message with realtime voicemail
Category: CDR/General
ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
* [f34a4d5a77] George Joseph -- cdr.c: Set event time on party b when
leaving a parking bridge
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to
deactivate CDR.
Reported by: Frederic LE FOLL
* [a0e31cc14e] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail
sometimes fails to deactivate CDR.
Category: Channels/chan_dahdi
ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times
out after 0:16:40
Reported by: Andrew Siplas
* [1c5116f842] Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to
better reflect "no timeout"
Category: Channels/chan_pjsip
ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group
endpoint/call_group
Reported by: Jean-Denis Girard
* [a0d1197ab3] Sean Bright -- res_pjsip_config_wizard: Fix change
detection for wizard settings
ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response
Contact
Reported by: Ross Beer
* [fd14d7302f] George Joseph -- res_pjsip_nat: Restore original contact
for REGISTER responses
Category: Channels/chan_sip/General
ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
Reported by: Jean Aunis - Prescom
* [811c7bdabe] Jean Aunis -- chan_sip: voice frames are no longer
transmitted after emitting a COLP
ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
Reported by: Jaco Kroon
* [32ce8fa046] Jaco Kroon -- chan_sip: in case of tcp/tls, be less
annoying about tx errors.
Category: Channels/chan_sip/Messaging
ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears
empty in the dialplan
Reported by: Frank Matano
* [7c137ebbd4] Sean Bright -- chan_sip.c: Stop handling continuation
lines after reading headers
Category: Channels/chan_sip/Transfers
ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
* [f34a4d5a77] George Joseph -- cdr.c: Set event time on party b when
leaving a parking bridge
Category: Codecs/codec_silk
ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation'
output
Reported by: Sean Bright
* [687890f734] Sean Bright -- translate.c: Fix silk 24kHz truncation in
'core show translation'
Category: Configs/Basic-PBX
ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order
mark is present
Reported by: Robin Leffmann
* [8c8989f205] Sean Bright -- config.c: Skip UTF-8 BOMs if present when
reading config files
Category: Contrib/General
ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed
to due to invalid syntax
Reported by: Richard Kenner
* [c4cd2fa7b9] snuffy -- contrib/valgrind: Fix use of frame-level
suppression
ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
Reported by: Pascal Cadotte Michaud
* [69c9b74836] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid
typo
Category: Core/Streams
ASTERISK-28625: Playback of local files impacted by large media cache
Reported by: Kevin Reeves
* [45f7f9bb85] Kevin Reeves -- main/file.c: Limit media cache usage to
remote files.
Category: Documentation
ASTERISK-24484: Update documentation for statsd module - usage
requirements unclear
Reported by: Dan Jenkins
* [deed0f5706] Sean Bright -- res_statsd: Document that res_statsd does
nothing on its own
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for
hostnames
Reported by: Joshua C. Colp
* [28275609b3] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
support for hostnames
ASTERISK-28507: Wiki docs missing for MessageWaiting
Reported by: David M. Lee
* [40bfd895a9] George Joseph -- CI: Update buildAsterisk.sh to do a
"make full"
Category: Functions/General
ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
Reported by: Pascal Cadotte Michaud
* [39aff53cff] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
argument documentation
Category: Functions/func_odbc
ASTERISK-28497: func_odbc: truncating Unicode string on readsql
Reported by: Boris P. Korzun
* [6d2eced3d3] Boris P. Korzun -- func_odbc: acf_odbc_read() and
cli_odbc_read() unicode support
Category: General
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
Reported by: Ted G
* [86822722b6] George Joseph -- res_rtp_asterisk: Add frame list
cleanups to ast_rtp_read
Category: PBX/General
ASTERISK-28695: core: minmemfree watermark uses free RAM, not available
RAM
Reported by: Kevin Flyn
* [9ff062f994] Sean Bright -- pbx.c: Include filesystem cache in free
memory calculation
ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent
command pri show span X
Reported by: Dirk Wendland
* [015cdc9f64] George Joseph -- sig_pri: Fix deadlock caused by
sig_pri_queue_hangup
Category: Resources/res_ari
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
* [19b0efce0d] Kevin Harwell -- res_stasis: trigger cleanup after update
Category: Resources/res_fax
ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config
option
Reported by: Kevin Harwell
* [68724425e4] Kevin Harwell -- res_fax: wrap v21 detected Asterisk
initiated negotiation with config option
Category: Resources/res_http_websocket
ASTERISK-28562: SIP WSS message not processed until next frame arrives
Reported by: Robert Sutton
* [f344356290] Sean Bright -- websocket: Consider pending SSL data when
waiting for socket input
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for
hostnames
Reported by: Joshua C. Colp
* [28275609b3] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
support for hostnames
Category: Resources/res_pjsip_notify
ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present
instead of just one
Reported by: AvayaXAsterisk
* [6898c8e0da] Sean Bright -- res_pjsip_notify: Only allow a single
Event header to be added to a NOTIFY
Category: Resources/res_pjsip_pubsub
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate
(ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
* [5b4d900e43] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence
data ref when recreating.
ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not
preserve XML version number
Reported by: Bryan Nelson
* [e17ba921f3] Joshua C. Colp -- res_pjsip_pubsub: Add ability to
persist generator state information.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
* [c9147a759e] Joshua C. Colp -- res_pjsip_session: Set stream state on
created streams for incoming SDP.
Category: Resources/res_pjsip_session
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
* [c9147a759e] Joshua C. Colp -- res_pjsip_session: Set stream state on
created streams for incoming SDP.
Category: Resources/res_realtime
ASTERISK-21794: CLI command 'realtime update2' syntax failure when using
according to usage help
Reported by: Cedric BASSAGET
* [3779e76b68] Sean Bright -- res_realtime: Fix 'realtime update2'
argument handling
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
* [78a907e23d] Kevin Harwell -- stasis/app: don't lock an app before a
call to send
ASTERISK-28633: stasis bridge topic leak
Reported by: Joeran Vinzens
* [e0449939ec] George Joseph -- stasis.c: Use correct topic name in
stasis_topic_pool_delete_topic
Category: Resources/res_statsd
ASTERISK-24484: Update documentation for statsd module - usage
requirements unclear
Reported by: Dan Jenkins
* [deed0f5706] Sean Bright -- res_statsd: Document that res_statsd does
nothing on its own
Improvement
Category: Applications/app_confbridge
ASTERISK-28658: app_confbridge: Add support for setting maximum sample
rate
Reported by: Joshua C. Colp
* [41fdcf7a65] Joshua C. Colp -- confbridge: Add support for specifying
maximum sample rate.
Category: Bridges/bridge_softmix
ASTERISK-28658: app_confbridge: Add support for setting maximum sample
rate
Reported by: Joshua C. Colp
* [41fdcf7a65] Joshua C. Colp -- confbridge: Add support for specifying
maximum sample rate.
Category: Channels/chan_pjsip
ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
Reported by: cmaj
* [62a8750d2d] Richard Mudgett -- app_chanisavail.c: Simplify dialplan
using ChanIsAvail.
* [b3d0b0c484] Richard Mudgett -- app_dial.c: Simplify dialplan using
Dial.
* [f7c6a7df9f] Richard Mudgett -- app_page.c: Simplify dialplan using
Page.
Category: Core/HTTP
ASTERISK-28710: Should be able to disable the /httpstatus URI in the
built-in HTTP server
Reported by: Sean Bright
* [1e79bf44fc] Sean Bright -- http: Add ability to disable /httpstatus
URI
Category: Documentation
ASTERISK-28673: GET FULL VARIABLE documentation clarification
Reported by: Jonathan Harris
* [7038c8676e] Sean Bright -- res_agi: Improve GET FULL VARIABLE
documentation
----------------------------------------------------------------------
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 |
|------------+-----------------+-----------------------------------------|
| 098c164ff7 | Joshua Colp | REVERT: Add option to suppress the |
| | | Message channel AMI and ARI events |
|------------+-----------------+-----------------------------------------|
| bdcf8e7165 | George Joseph | message.c: Add option to suppress the |
| | | Message channel AMI and ARI events |
|------------+-----------------+-----------------------------------------|
| | Asterisk | |
| 8b027e2490 | Development | Update for 17.2.0-rc2 |
| | Team | |
|------------+-----------------+-----------------------------------------|
| | Asterisk | |
| 4579ddd846 | Development | Update for 17.2.0-rc1 |
| | Team | |
|------------+-----------------+-----------------------------------------|
| | Asterisk | Update CHANGES and UPGRADE.txt for |
| a0f44b1052 | Development | 17.2.0 |
| | Team | |
|------------+-----------------+-----------------------------------------|
| d33cd39260 | Sean Bright | func_odbc.conf.sample: Add example |
| | | lookup |
|------------+-----------------+-----------------------------------------|
| 7b32763190 | Rodrigo RamÃrez | queue_log: Add alembic script for |
| | Norambuena | generate db table for queue_log |
|------------+-----------------+-----------------------------------------|
| cb6f106d76 | Sean Bright | app_voicemail, say: Fix various leading |
| | | whitespace problems |
|------------+-----------------+-----------------------------------------|
| | | netsock2: |
| 0a9e5febd5 | Jaco Kroon | ast_addressfamily_to_sockaddrsize and |
| | | ast_sockaddr_from_sockaddr. |
|------------+-----------------+-----------------------------------------|
| 21d8d2426d | Kevin Harwell | app_agent_pool: Update XML docs for |
| | | AgentLogin |
|------------+-----------------+-----------------------------------------|
| 919da21690 | Richard Mudgett | features.c: Make Bridge application |
| | | tolerate unspecified channel. |
|------------+-----------------+-----------------------------------------|
| a82fc35606 | Richard Mudgett | app_chanspy.c: Reduce log message level |
| | | from notice to verbose. |
|------------+-----------------+-----------------------------------------|
| 4dfa19eb27 | Richard Mudgett | app_softhangup.c: Reduce unnecessary |
| | | warning to verbose message. |
|------------+-----------------+-----------------------------------------|
| 843d601e2c | Sean Bright | db: Initialize condition primitive |
| | | before use |
|------------+-----------------+-----------------------------------------|
| 6041fc8b9c | Jaco Kroon | ACL: ast_apply_acl_nolog - identical to |
| | | ast_apply_acl but without logging. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
asterisk-17.1.0-summary.html | 372 ---
asterisk-17.1.0-summary.txt | 962 ----------
b/.version | 2
b/CHANGES | 55
b/ChangeLog | 834 ++++++++
b/apps/app_agent_pool.c | 4
b/apps/app_chanisavail.c | 140 -
b/apps/app_chanspy.c | 3
b/apps/app_confbridge.c | 2
b/apps/app_dial.c | 51
b/apps/app_page.c | 30
b/apps/app_queue.c | 2
b/apps/app_record.c | 3
b/apps/app_softhangup.c | 2
b/apps/app_voicemail.c | 407 ++--
b/apps/confbridge/conf_config_parser.c | 17
b/apps/confbridge/include/confbridge.h | 1
b/asterisk-17.2.0-rc2-summary.html | 15
b/asterisk-17.2.0-rc2-summary.txt | 96
b/bridges/bridge_softmix.c | 18
b/channels/chan_dahdi.c | 2
b/channels/chan_sip.c | 30
b/channels/sig_pri.c | 23
b/configs/samples/confbridge.conf.sample | 4
b/configs/samples/func_odbc.conf.sample | 8
b/configs/samples/http.conf.sample | 10
b/configs/samples/pjsip.conf.sample | 1
b/contrib/ast-db-manage/README.md | 1
b/contrib/ast-db-manage/queue_log.ini.sample | 58
b/contrib/ast-db-manage/queue_log/env.py | 1
b/contrib/ast-db-manage/queue_log/script.py.mako | 24
b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py | 38
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/contrib/valgrind.supp | 14
b/doc/CHANGES-staging/res_fax_negotiate_both | 7
b/doc/appdocsxml.dtd | 2
b/funcs/func_curl.c | 11
b/funcs/func_odbc.c | 22
b/funcs/func_pjsip_contact.c | 6
b/include/asterisk/acl.h | 37
b/include/asterisk/bridge.h | 21
b/include/asterisk/http_websocket.h | 14
b/include/asterisk/iostream.h | 14
b/include/asterisk/netsock2.h | 42
b/include/asterisk/res_fax.h | 3
b/include/asterisk/res_pjsip_pubsub.h | 23
b/include/asterisk/stasis.h | 3
b/main/acl.c | 74
b/main/bridge.c | 7
b/main/cdr.c | 15
b/main/config.c | 12
b/main/db.c | 3
b/main/features.c | 28
b/main/file.c | 7
b/main/http.c | 56
b/main/iostream.c | 14
b/main/pbx.c | 12
b/main/say.c | 956 ++++-----
b/main/stasis.c | 17
b/main/translate.c | 8
b/res/res_agi.c | 20
b/res/res_fax.c | 26
b/res/res_http_websocket.c | 11
b/res/res_pjsip/pjsip_message_filter.c | 40
b/res/res_pjsip_config_wizard.c | 7
b/res/res_pjsip_dialog_info_body_generator.c | 80
b/res/res_pjsip_endpoint_identifier_ip.c | 86
b/res/res_pjsip_nat.c | 84
b/res/res_pjsip_notify.c | 22
b/res/res_pjsip_pubsub.c | 87
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_realtime.c | 56
b/res/res_stasis.c | 8
b/res/res_statsd.c | 35
74 files changed, 2848 insertions(+), 2362 deletions(-)