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Update for 13.17.1

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Kevin Harwell
2017-08-31 10:44:55 -05:00
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13.17.0
13.17.1

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2017-08-31 15:44 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.17.1 Released.
2017-07-24 10:48 +0000 [0e5b7743d9] George Joseph <gjoseph@digium.com>
* pjsip_message_ip_updater: Fix issue handling "tel" URIs
sanitize_tdata was assuming all URIs were SIP URIs so when a non
SIP uri was in the From, To or Contact headers, the unconditional
cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
a segfault when trying to access uri->other_param.
* Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
checks before attempting to cast or use the returned uri.
ASTERISK-27152
Reported-by: Ross Beer
Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
2017-07-01 19:24 +0000 [707892089d] Corey Farrell <git@cfware.com>
* AST-2017-006: Fix app_minivm application MinivmNotify command injection
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
ASTERISK-27103
Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
2017-05-22 10:36 +0000 [3ee5c6dcbe] Joshua Colp <jcolp@digium.com>
* res_rtp_asterisk: Only learn a new source in learn state.
This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.
The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.
Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.
ASTERISK-27013
Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
2017-07-12 11:12 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.17.0 Released.

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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-13.17.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.17.0</h3><h3 align="center">Date: 2017-07-12</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#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-13.16.0.</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%">17 Sean Bright <sean.bright@gmail.com><br/>12 George Joseph <gjoseph@digium.com><br/>10 Joshua Colp <jcolp@digium.com><br/>9 Alexei Gradinari <alex2grad@gmail.com><br/>5 Richard Mudgett <rmudgett@digium.com><br/>5 Kevin Harwell <kharwell@digium.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>1 Jan Friesse <jfriesse@redhat.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/>1 David M. Lee <dlee@digium.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Rodrigo Ramirez Norambuena <a@rodrigoramirez.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Corey Farrell <git@cfware.com><br/></td><td width="33%"><td width="33%">4 Alexei Gradinari <alex2grad@gmail.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Louis Jocelyn Paquet <ljpaquet@quebecinternet.net><br/>3 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>3 George Joseph <gjoseph@digium.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Michael Walton <mike@farsouthnet.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Jacek Konieczny <jkonieczny@eggsoft.pl><br/>1 Tim Morgan <morganuci@gmail.com><br/>1 Etienne Allovon <eallovon@avencall.com><br/>1 alex <asterisk@maximum.guru><br/>1 Kinsey Moore <kmoore@digium.com><br/>1 John Harris <john.harris@certus-tech.com><br/>1 Javier Riveros <goseeped@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Chris Howard <choward@digium.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Andrew Nowrot <andrew.nowrot@gmail.com><br/>1 'alex'<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Ben Langfeld <ben@langfeld.me><br/>1 John Fawcett <john@voipsupport.it><br/>1 Corey Farrell <git@cfware.com><br/>1 Frankie Chin <fchin@biamp.com><br/>1 Zach R <zrothy@monmouth.com><br/>1 Matthias Binder <it@mitterhuemer.at><br/>1 Christopher van de Sande <cvandesande@opendmz.com><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Antoine Pitrou <pitrou@free.fr><br/>1 Alex <metsys@gmx.com><br/>1 Etienne Lessard <elessard97@gmail.com><br/>1 Ryan Smith <ryan.smith@tekara.co.uk><br/>1 Michael Maier <m1278468@mailbox.org><br/>1 OpenBSD ports<br/>1 Marek Cervenka <marek.cervenka@gmail.com><br/>1 Ronald Raikes <reraikes@avweb.com><br/>1 Ove Aursand <oveaurs@gmail.com><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 wushumasters <wushumasters@gmail.com><br/>1 Tony Mountifield <tony@softins.co.uk><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Michel R. Vaillancourt <michel@jkl5group.com><br/>1 David Brillert <david_brillert@scopserv.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><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/format_mp3</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23951">ASTERISK-23951</a>: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97b003f5e2d4a350508fc20173e180a23f8ef525">[97b003f5e2]</a> Sean Bright -- format_mp3: Re-work menuselect/build issues</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72213c98e3d4d5287ed321f1b4fb67087a7a129c">[72213c98e3]</a> Sean Bright -- format_mp3: Don't try to build format_mp3 if we don't have sources</li>
</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27012">ASTERISK-27012</a>: app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br/>Reported by: Robert Mordec<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1b32de2c5fb8854183f0c7d8c9df7470ab9c140">[f1b32de2c5]</a> Robert Mordec -- app_confbridge: Race between removing and playing name recording while leaving</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25665">ASTERISK-25665</a>: Duplicate logging in queue log for EXITEMPTY events<br/>Reported by: Ove Aursand<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27065">ASTERISK-27065</a>: call hangup after leaving app_queue<br/>Reported by: Marek Cervenka<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26400">ASTERISK-26400</a>: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26715">ASTERISK-26715</a>: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel<br/>Reported by: David Brillert<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26975">ASTERISK-26975</a>: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call<br/>Reported by: Lorne Gaetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24052">ASTERISK-24052</a>: app_voicemail reloads result in leaked IMAP sockets.<br/>Reported by: Louis Jocelyn Paquet<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b6c327c515944d74aa798f385e01768a4bb04c2">[3b6c327c51]</a> Alexei Gradinari -- app_voicemail: IMAP logout on reload/unload</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08be5e01e8ab72a7e9e80525e20967467a6df99b">[08be5e01e8]</a> Alexei Gradinari -- app_voicemail: IMAP logout on MWI unsubscribe</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27039">ASTERISK-27039</a>: chan_pjsip: Device state is idle when channel from endpoint is in early media<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f10c6b3b044f9979c523f65f449670047dcb57f">[1f10c6b3b0]</a> Joshua Colp -- chan_pjsip: Update device state when in early media.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26996">ASTERISK-26996</a>: chan_pjsip: Flipping between codecs<br/>Reported by: Michael Maier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=996a4791ff123e80d71d44cb0fd13bb201d197b1">[996a4791ff]</a> Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26281">ASTERISK-26281</a>: chan_pjsip would send INVITE to 'Unreachable' endpoints<br/>Reported by: Jacek Konieczny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=746c2c574578608a6b48d4794ba33cda5a6dd484">[746c2c5745]</a> Joshua Colp -- res_pjsip: Add support for returning only reachable contacts and use it.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27106">ASTERISK-27106</a>: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39d2ebbf56635355432eb96ff850c0c9bf2a5d63">[39d2ebbf56]</a> Alexander Traud -- chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9f4b3b966e911fae157a484d8f4a1440130eede6">[9f4b3b966e]</a> Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4479038073e57a67c19c1ec5dc8896fcc8c3a0fb">[4479038073]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
</ul><br><h4>Category: Channels/chan_sip/SRTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27075">ASTERISK-27075</a>: bridge: stuck channel(s) after failed attended transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=67664fbf95a00ced30f8791fd1089b4595e29479">[67664fbf95]</a> Kevin Harwell -- bridge: stuck channel(s) after failed attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27100">ASTERISK-27100</a>: channel: ast_waitfordigit_full fails to clear flag in an error branch.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73520e9f58857049a086fb88106e342cdc25d3a1">[73520e9f58]</a> Corey Farrell -- channel: Clear channel flag in error branch.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27074">ASTERISK-27074</a>: core_local: local channel data not being properly unref'ed and unlocked<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f9913f2723cbcbf6d78f4da7ee4dd4decc13c05">[1f9913f272]</a> Kevin Harwell -- core_local: local channel data not being properly unref'ed and unlocked</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26789">ASTERISK-26789</a>: Audit manipulation of channel flags without locks<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=161820396495a549c9a378d32136cbb5f28ef2af">[1618203964]</a> Joshua Colp -- asterisk: Audit locking of channel when manipulating flags.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27041">ASTERISK-27041</a>: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc307af7f2ed653914aeadb0b7e613cb4e239b06">[dc307af7f2]</a> Frederic LE FOLL -- Core/PBX: Deadlock between dialplan execution and application unregistration.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26978">ASTERISK-26978</a>: rtp: Crash in ast_rtp_codecs_payload_code()<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb48e99bd4f4556424a6799e2e5f7aebf8911e8d">[eb48e99bd4]</a> George Joseph -- bridge_native_rtp: Keep rtp instance refs on bridge_channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27057">ASTERISK-27057</a>: Seg Fault in ast_sorcery_object_get_id at sorcery.c<br/>Reported by: Ryan Smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2eea791e4178e5f2e4446a5f70d81ac27cf2a0e">[c2eea791e4]</a> George Joseph -- res_pjsip_pubsub: Fix reference to released endpoint</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839">ASTERISK-23839</a>: AGI - RECORD FILE - documentation doesn't describe BEEP argument<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3eb7fbba72482b3019a7493c68e533e67d9d8235">[3eb7fbba72]</a> Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27108">ASTERISK-27108</a>: Crash using 'data get' CLI command<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6258de458b2e6ba02e91ed67bbd2801f0984526a">[6258de458b]</a> Sean Bright -- core: Fix segfault when invoking 'data get' CLI command</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27060">ASTERISK-27060</a>: Comment typo format_g729.c<br/>Reported by: Matthew Fredrickson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a40073750b46ae28ddf1041d5ed3ab57151298e">[0a40073750]</a> Matthew Fredrickson -- formats/format_g729: Fix typo in comment</li>
</ul><br><h4>Category: PBX/pbx_realtime</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19291">ASTERISK-19291</a>: Background in realtime<br/>Reported by: Andrew Nowrot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=283cc59af746896a2b2bc23899fc86118895f7c0">[283cc59af7]</a> Sean Bright -- pbx_builtin: Properly handle hangup during Background</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839">ASTERISK-23839</a>: AGI - RECORD FILE - documentation doesn't describe BEEP argument<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3eb7fbba72482b3019a7493c68e533e67d9d8235">[3eb7fbba72]</a> Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22432">ASTERISK-22432</a>: Async AGI crashes Asterisk when issuing "set variable" command without args<br/>Reported by: Antoine Pitrou<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f306e451f6f905a2bb74c15cb844735c244a7610">[f306e451f6]</a> Sean Bright -- res_agi: Prevent crash when SET VARIABLE called without arguments</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25662">ASTERISK-25662</a>: Malformed AGI 520 Usage response<br/>Reported by: Tony Mountifield<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a007e438c36960d4179e2f188767e7ae14a204d1">[a007e438c3]</a> Sean Bright -- res_agi: Fix malformed AGI usage response</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27026">ASTERISK-27026</a>: res_ari: Crash when no ari.conf configuration file exists<br/>Reported by: Ronald Raikes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7901b9853e8f60e1d2dce44ce81dec6f7f866ccc">[7901b9853e]</a> George Joseph -- res_ari: Add "module loaded" check to ari stubs</li>
</ul><br><h4>Category: Resources/res_ari_recordings</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27021">ASTERISK-27021</a>: GET /recordings/stored returns 500 Internal Server Error<br/>Reported by: Tim Morgan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf6cf59646f52dc3de12dac16c3c3824ce9ae927">[cf6cf59646]</a> Sean Bright -- stasis_recording: Correct ast_asprintf error checking</li>
</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27008">ASTERISK-27008</a>: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br/>Reported by: John Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=700ef6861ab966008ca16e5f23c64eb68b047c08">[700ef6861a]</a> Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp attributes</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27090">ASTERISK-27090</a>: PJSIP: Deadlock using TCP transport<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d64cbde5756eaa1c7ee62116e112b7ebd198bbe">[0d64cbde57]</a> Richard Mudgett -- pjsip_distributor.c: Fix deadlock with TCP type transports.</li>
</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27052">ASTERISK-27052</a>: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br/>Reported by: alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0bde568669ac26735c1058115ae96223a7e69a6b">[0bde568669]</a> George Joseph -- pjproject_bundled: Use the asterisk github mirror for download</li>
</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053">ASTERISK-27053</a>: res_pjsip_refer/session: Calls dropped during transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27024">ASTERISK-27024</a>: nat/external_media settings ignored in 14.4.1<br/>Reported by: Christopher van de Sande<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2dee95cc7a280d0ab84c778bf44a76aa62ac758b">[2dee95cc7a]</a> Florian Floimair -- res_pjsip_session: Correct inverted test in session_outgoing_nat_hook</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053">ASTERISK-27053</a>: res_pjsip_refer/session: Calls dropped during transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26964">ASTERISK-26964</a>: res_pjsip_session: Wrong From on reinvite when request and To URI differ<br/>Reported by: Yasin CANER<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36628cc9c474b52b134a415803b14f87e420dce6">[36628cc9c4]</a> Yasin CANER -- res_pjsip_session : fixed wrong From Header number On Re-invite</li>
</ul><br><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27046">ASTERISK-27046</a>: res_pjsip_transport_websocket: segfault in get_write_timeout<br/>Reported by: Jørgen H<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e16a669c70c5a93bb9a38c218a5348cd62bd780a">[e16a669c70]</a> Jørgen H -- res_pjsip_transport_websocket: Add NULL check in get_write_timeout</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27022">ASTERISK-27022</a>: res_rtp_asterisk: Incorrect SSRC change for RTCP component<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7dafe82751fd512d58bb3843601daff013958dd2">[7dafe82751]</a> George Joseph -- res_rtp_asterisk: Fix ssrc change for rtcp srtp</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4479038073e57a67c19c1ec5dc8896fcc8c3a0fb">[4479038073]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25294">ASTERISK-25294</a>: srtp's crypto_get_random deprecated<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
</ul><br><h4>Category: Resources/res_stasis_snoop</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26333">ASTERISK-26333</a>: Problems with Blind Transfer, PJSIP (Aastra 6869i)<br/>Reported by: Matthias Binder<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6af2dd34afc2c20bdabd07bc3836821690db4c86">[6af2dd34af]</a> Alexei Gradinari -- res_pjsip: New endpoint option "refer_blind_progress"</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27043">ASTERISK-27043</a>: Core/BuildSystem: Add defines to fix build with LibreSSL<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a64f65fe6fee96702668bdd3344233f19232850">[6a64f65fe6]</a> Guido Falsi -- BuildSystem: Add patches to allow building with recent LibreSSL</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419">ASTERISK-26419</a>: audiohooks: Remove redundant codec translations when using audiohooks<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419">ASTERISK-26419</a>: audiohooks: Remove redundant codec translations when using audiohooks<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Core/Portability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27042">ASTERISK-27042</a>: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44cee2f4a15db911d2c9bdd6f845d17a1e6c6c17">[44cee2f4a1]</a> Guido Falsi -- BuildSystem: Fix build on FreeBSD due to missing crypt.h</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26124">ASTERISK-26124</a>: res_agi: Set audio format for EAGI audio stream<br/>Reported by: John Fawcett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90237dca11d0adf129198cef4a6a0716a52618b5">[90237dca11]</a> Sean Bright -- res_agi: Allow configuration of audio format of EAGI pipe</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26230">ASTERISK-26230</a>: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f6a9617eb44a8d59b5828cd860d3852cc824ce9">[0f6a9617eb]</a> Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59c9bbe6961a5677ddb13eed2a130d16b6ffc0ee">[59c9bbe696]</a> Alexei Gradinari -- res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26976">ASTERISK-26976</a>: libsrtp-2.x.x support<br/>Reported by: Alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</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: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26469">ASTERISK-26469</a>: Infinite loop after a dual Redirect<br/>Reported by: Etienne Allovon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b07b2162359ccc9a3f84324fabce18b6ad63eee3">[b07b216235]</a> Joshua Colp -- manager: Clear the flag on the other channel.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27095">ASTERISK-27095</a>: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bd7c0f37cb7b513d1333717ece0118bd8875546">[6bd7c0f37c]</a> George Joseph -- chan_pjsip: Fix ability to send UPDATE on COLP</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27016">ASTERISK-27016</a>: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.<br/>Reported by: Chris Howard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4910a3bf402baddf8ed72badfaed7ae64da48686">[4910a3bf40]</a> Joshua Colp -- channel: Fix reference counting in ast_channel_suppress.</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27088">ASTERISK-27088</a>: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0426b1d88ab97c4fc1b2b27f8da93b28096f2dfc">[0426b1d88a]</a> Joshua Colp -- res_rtp_asterisk: Fix issues with ICE renegotiation.</li>
</ul><br><h4>Category: Resources/res_corosync</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25370">ASTERISK-25370</a>: res_corosync segfaults at startup with corosync version > 2.x<br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=005a4afa6b0e710e11b47b11cfc152b028c596fc">[005a4afa6b]</a> Jan Friesse -- res_corosync: Change thread stack size</li>
</ul><br><h4>Category: Resources/res_pjsip_dialog_info_body_generator</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26919">ASTERISK-26919</a>: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip<br/>Reported by: Zach R<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6e4899612ca71bc3c9180dadea0c0117e8ae462">[a6e4899612]</a> Alexei Gradinari -- res_pjsip: New endpoint option "notify_early_inuse_ringing"</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27051">ASTERISK-27051</a>: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e749c8f51c20fb13bfe93e969cf02d7e74cdb27">[8e749c8f51]</a> Alexei Gradinari -- res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27059">ASTERISK-27059</a>: res_stasis: Stolen channel references are leaking<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=edfdb4dff5d8438bdb1dfb526c57618944ea6bf3">[edfdb4dff5]</a> George Joseph -- res_stasis: Plug reference leak on stolen channels</li>
</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27097">ASTERISK-27097</a>: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbe68f139db525b2d922f63d8452d9732fb5f1b9">[bbe68f139d]</a> George Joseph -- pjproject_bundled: Allow passing configure options to bundled</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27068">ASTERISK-27068</a>: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fbc34d2bd5393d93d8b3b3a8c6daa895c2e9633">[9fbc34d2bd]</a> Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fbc34d2bd5393d93d8b3b3a8c6daa895c2e9633">[9fbc34d2bd]</a> Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode</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=0c00ee754b436ca926b92b469ce259e8fdc8732e">0c00ee754b</a></td><td>George Joseph</td><td>Update for 13.17.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=379fe658312e11699ff8c8e8a463e31b3c277237">379fe65831</a></td><td>George Joseph</td><td>Fix alembic branches</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=905d18e8bf52ea7657acaaf2ec0cbe58531fb625">905d18e8bf</a></td><td>Richard Mudgett</td><td>pjsip_distributor.c: Fix unidentified_requests hash functions.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f59d08924bc676970cabc6f3e291c7d1d2f2707">1f59d08924</a></td><td>Torrey Searle</td><td>res/res_pjsip_t38: fix incorrect increment of media_count</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=764d04fa8705d9e5c2e7aee8a6f1c774d7d28595">764d04fa87</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cecf6540dc4779598289f711340bb966bbfcc6aa">cecf6540dc</a></td><td>Rodrigo Ramírez Norambuena</td><td>cdr: fix mistake spelling of a word for Unanswered.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9a4ab8c8c00c8d53584d6f7e31729b5027c8dd6">b9a4ab8c8c</a></td><td>Richard Mudgett</td><td>chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1a209d5ac8f8b7fe96e54d6aba55dbf0dbb1403">f1a209d5ac</a></td><td>Richard Mudgett</td><td>app_voicemail.c: Fix compile error when IMAP enabled.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68de35a6a01e2a1fe732e156b73f800bb672a421">68de35a6a0</a></td><td>David M. Lee</td><td>CFLAGS for BIND8 support</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da3312457e6cf1c0d7bc8cb2a4aba57877fb5afc">da3312457e</a></td><td>Sean Bright</td><td>codecs.conf.sample: Fix max_bandwidth speling error</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=590ffcaf0b03bbe3d25730ad750a2075a46c7208">590ffcaf0b</a></td><td>Sean Bright</td><td>eventfd: Disable during cross compilation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5520b6c201875133a73db5a2c88b5fc5b78864bb">5520b6c201</a></td><td>Alexei Gradinari</td><td>CHANGES: correct version for a new option 'refer_blind_progress'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c093bf8072ff65bf29d290c1330291c460cd7fdf">c093bf8072</a></td><td>Sean Bright</td><td>res_rtp_multicast: Use consistent timestamps when possible</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c10341646d353922b4ee92c77fc4e5560d263c73">c10341646d</a></td><td>George Joseph</td><td>test_json: Fix test names with reserved words</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=65898c3af82e2d780a48d9d50d3b1c952c208a89">65898c3af8</a></td><td>George Joseph</td><td>unittests: Add a unit test that causes a SEGV and...</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-13.16.0-summary.html | 405 ---
asterisk-13.16.0-summary.txt | 952 ---------
b/.version | 2
b/CHANGES | 54
b/ChangeLog | 1045 +++++++++-
b/Makefile | 3
b/addons/Makefile | 10
b/apps/app_chanspy.c | 16
b/apps/app_confbridge.c | 79
b/apps/app_dial.c | 6
b/apps/app_disa.c | 10
b/apps/app_dumpchan.c | 4
b/apps/app_externalivr.c | 6
b/apps/app_meetme.c | 2
b/apps/app_queue.c | 109 -
b/apps/app_voicemail.c | 80
b/asterisk-13.17.0-rc1-summary.html | 311 ++
b/asterisk-13.17.0-rc1-summary.txt | 832 +++++++
b/autoconf/ast_ext_lib.m4 | 36
b/bridges/bridge_native_rtp.c | 677 +++++-
b/bridges/bridge_simple.c | 32
b/channels/chan_pjsip.c | 68
b/channels/chan_sip.c | 8
b/channels/pjsip/dialplan_functions.c | 37
b/configs/samples/cdr.conf.sample | 2
b/configs/samples/codecs.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 20
b/configs/samples/sip.conf.sample | 3
b/configs/samples/voicemail.conf.sample | 3
b/configure | 434 +++-
b/configure.ac | 100
b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
b/contrib/realtime/mssql/mssql_config.sql | 46
b/contrib/realtime/mysql/mysql_config.sql | 18
b/contrib/realtime/oracle/oracle_config.sql | 46
b/contrib/realtime/postgresql/postgresql_config.sql | 22
b/formats/format_g729.c | 2
b/include/asterisk/ari.h | 10
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/bridge_channel.h | 2
b/include/asterisk/bridge_channel_internal.h | 11
b/include/asterisk/bridge_technology.h | 3
b/include/asterisk/channel.h | 25
b/include/asterisk/codec.h | 3
b/include/asterisk/core_local.h | 37
b/include/asterisk/format.h | 11
b/include/asterisk/res_pjsip.h | 74
b/include/asterisk/res_pjsip_presence_xml.h | 3
b/include/asterisk/res_pjsip_session.h | 11
b/include/asterisk/rtp_engine.h | 9
b/include/asterisk/smoother.h | 1
b/include/asterisk/test.h | 8
b/main/autoservice.c | 2
b/main/bridge.c | 10
b/main/bridge_after.c | 2
b/main/bridge_channel.c | 38
b/main/channel.c | 90
b/main/codec_builtin.c | 19
b/main/core_local.c | 54
b/main/crypt.c | 2
b/main/data.c | 4
b/main/file.c | 20
b/main/format.c | 8
b/main/libasteriskssl.c | 4
b/main/manager.c | 8
b/main/pbx.c | 4
b/main/pbx_app.c | 7
b/main/pbx_builtins.c | 8
b/main/tcptls.c | 4
b/main/test.c | 4
b/makeopts.in | 2
b/res/res_agi.c | 73
b/res/res_ari_applications.c | 4
b/res/res_ari_asterisk.c | 4
b/res/res_ari_bridges.c | 4
b/res/res_ari_channels.c | 4
b/res/res_ari_device_states.c | 4
b/res/res_ari_endpoints.c | 4
b/res/res_ari_events.c | 33
b/res/res_ari_mailboxes.c | 4
b/res/res_ari_playbacks.c | 4
b/res/res_ari_recordings.c | 4
b/res/res_ari_sounds.c | 4
b/res/res_corosync.c | 29
b/res/res_format_attr_h263.c | 2
b/res/res_format_attr_h264.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip.c | 31
b/res/res_pjsip/location.c | 53
b/res/res_pjsip/pjsip_configuration.c | 9
b/res/res_pjsip/pjsip_distributor.c | 242 +-
b/res/res_pjsip/presence_xml.c | 9
b/res/res_pjsip_dialog_info_body_generator.c | 10
b/res/res_pjsip_mwi.c | 87
b/res/res_pjsip_pidf_body_generator.c | 2
b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 8
b/res/res_pjsip_refer.c | 28
b/res/res_pjsip_sdp_rtp.c | 38
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_session.exports.in | 1
b/res/res_pjsip_t38.c | 2
b/res/res_pjsip_transport_websocket.c | 4
b/res/res_pjsip_xpidf_body_generator.c | 2
b/res/res_rtp_asterisk.c | 41
b/res/res_rtp_multicast.c | 139 +
b/res/res_srtp.c | 15
b/res/res_stasis.c | 20
b/res/srtp/srtp_compat.h | 29
b/res/stasis_recording/stored.c | 4
b/rest-api-templates/res_ari_resource.c.mustache | 35
b/tests/test_bridging.c | 292 ++
b/tests/test_json.c | 16
b/tests/test_pbx.c | 22
b/third-party/configure.m4 | 5
b/third-party/pjproject/Makefile | 2
b/third-party/pjproject/Makefile.rules | 7
b/third-party/pjproject/configure.m4 | 24
b/third-party/pjproject/patches/0070-Set-PJSIP_INV_SUPPORT_UPDATE-correctly-in-pjsip_inv_.patch | 16
121 files changed, 5477 insertions(+), 2043 deletions(-)</pre><br></html>

814
asterisk-13.17.0-summary.txt
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@@ -1,814 +0,0 @@
Release Summary
asterisk-13.17.0
Date: 2017-07-12
<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-13.16.0.
----------------------------------------------------------------------
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
17 Sean Bright 4 Alexei Gradinari
12 George Joseph 4 Joshua Colp
10 Joshua Colp 3 Kevin Harwell
9 Alexei Gradinari 3 Louis Jocelyn Paquet
5 Richard Mudgett 3 Tzafrir Cohen
5 Kevin Harwell 3 George Joseph
2 Torrey Searle 2 Guido Falsi
2 Guido Falsi 2 Alexander Traud
2 Alexander Traud 2 Michael Walton
1 Jan Friesse 2 Torrey Searle
1 Florian Floimair 1 Rusty Newton
1 Ivan Poddubny 1 Matthew Fredrickson
1 Matthew Fredrickson 1 Jacek Konieczny
1 Yasin CANER 1 Tim Morgan
1 David M. Lee 1 Etienne Allovon
1 Robert Mordec 1 alex
1 JA,rgen H 1 Kinsey Moore
1 Rodrigo Ramirez Norambuena 1 John Harris
1 Frederic LE FOLL 1 Javier Riveros
1 Corey Farrell 1 Sean Bright
1 Robert Mordec
1 Ross Beer
1 Chris Howard
1 mdu113
1 Andrew Nowrot
1 'alex'
1 Lorne Gaetz
1 Ben Langfeld
1 John Fawcett
1 Corey Farrell
1 Frankie Chin
1 Zach R
1 Matthias Binder
1 Christopher van de Sande
1 Stefan EngstrAP:m
1 Antoine Pitrou
1 Alex
1 Etienne Lessard
1 Ryan Smith
1 Michael Maier
1 OpenBSD ports
1 Marek Cervenka
1 Ronald Raikes
1 Ove Aursand
1 Richard Mudgett
1 Frederic LE FOLL
1 wushumasters
1 Tony Mountifield
1 JA,rgen H
1 Michel R. Vaillancourt
1 David Brillert
1 Yasin CANER
----------------------------------------------------------------------
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/format_mp3
ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if
mp3lib was not downloaded
Reported by: Tzafrir Cohen
* [97b003f5e2] Sean Bright -- format_mp3: Re-work menuselect/build
issues
* [72213c98e3] Sean Bright -- format_mp3: Don't try to build format_mp3
if we don't have sources
Category: Applications/app_confbridge
ASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user
name recording while leaving
Reported by: Robert Mordec
* [f1b32de2c5] Robert Mordec -- app_confbridge: Race between removing
and playing name recording while leaving
Category: Applications/app_meetme
ASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua Colp
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
Category: Applications/app_queue
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove Aursand
* [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
when a queue is empty
ASTERISK-27065: call hangup after leaving app_queue
Reported by: Marek Cervenka
* [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
when a queue is empty
ASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumasters
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26400: app_queue: Queue member stops being called after AMI
"Redirect" action for queues with wrapuptime
Reported by: Etienne Lessard
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26715: app_queue: Member will not receive any new calls after
doing a transfer if wrapuptime = greater than 0 and using Local channel
Reported by: David Brillert
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to
receive queue calls after transfer queue call
Reported by: Lorne Gaetz
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
Category: Applications/app_voicemail/IMAP
ASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets.
Reported by: Louis Jocelyn Paquet
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
control
* [3b6c327c51] Alexei Gradinari -- app_voicemail: IMAP logout on
reload/unload
* [08be5e01e8] Alexei Gradinari -- app_voicemail: IMAP logout on MWI
unsubscribe
Category: Bridges/bridge_simple
ASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. Vaillancourt
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Channels/chan_pjsip
ASTERISK-27039: chan_pjsip: Device state is idle when channel from
endpoint is in early media
Reported by: Joshua Colp
* [1f10c6b3b0] Joshua Colp -- chan_pjsip: Update device state when in
early media.
ASTERISK-26996: chan_pjsip: Flipping between codecs
Reported by: Michael Maier
* [996a4791ff] Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec'
option to include us changing.
ASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints
Reported by: Jacek Konieczny
* [746c2c5745] Joshua Colp -- res_pjsip: Add support for returning only
reachable contacts and use it.
Category: Channels/chan_sip/General
ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really
different domain with TLS.
Reported by: Alexander Traud
* [39d2ebbf56] Alexander Traud -- chan_sip: Only when different, add
TCP|TLS in autodomain (SIP Domain Support).
* [9f4b3b966e] Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr
in autodomain (SIP Domain Support).
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngstrAP:m
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
Category: Channels/chan_sip/SRTP
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
Category: Codecs/General
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
Category: Core/Bridging
ASTERISK-27075: bridge: stuck channel(s) after failed attended transfer
Reported by: Kevin Harwell
* [67664fbf95] Kevin Harwell -- bridge: stuck channel(s) after failed
attended transfer
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
bridge
Reported by: Torrey Searle
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
Category: Core/Channels
ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an
error branch.
Reported by: Corey Farrell
* [73520e9f58] Corey Farrell -- channel: Clear channel flag in error
branch.
ASTERISK-27074: core_local: local channel data not being properly unref'ed
and unlocked
Reported by: Kevin Harwell
* [1f9913f272] Kevin Harwell -- core_local: local channel data not being
properly unref'ed and unlocked
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
bridge
Reported by: Torrey Searle
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
ASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua Colp
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
Category: Core/General
ASTERISK-26789: Audit manipulation of channel flags without locks
Reported by: Joshua Colp
* [1618203964] Joshua Colp -- asterisk: Audit locking of channel when
manipulating flags.
Category: Core/PBX
ASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and
application unregistration
Reported by: Frederic LE FOLL
* [dc307af7f2] Frederic LE FOLL -- Core/PBX: Deadlock between dialplan
execution and application unregistration.
Category: Core/RTP
ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code()
Reported by: Ross Beer
* [eb48e99bd4] George Joseph -- bridge_native_rtp: Keep rtp instance
refs on bridge_channel
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
Category: Core/Sorcery
ASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c
Reported by: Ryan Smith
* [c2eea791e4] George Joseph -- res_pjsip_pubsub: Fix reference to
released endpoint
Category: Documentation
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
argument
Reported by: Rusty Newton
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
documentation
Category: General
ASTERISK-27108: Crash using 'data get' CLI command
Reported by: Sean Bright
* [6258de458b] Sean Bright -- core: Fix segfault when invoking 'data
get' CLI command
ASTERISK-27060: Comment typo format_g729.c
Reported by: Matthew Fredrickson
* [0a40073750] Matthew Fredrickson -- formats/format_g729: Fix typo in
comment
Category: PBX/pbx_realtime
ASTERISK-19291: Background in realtime
Reported by: Andrew Nowrot
* [283cc59af7] Sean Bright -- pbx_builtin: Properly handle hangup during
Background
Category: Resources/res_agi
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
argument
Reported by: Rusty Newton
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
documentation
ASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable"
command without args
Reported by: Antoine Pitrou
* [f306e451f6] Sean Bright -- res_agi: Prevent crash when SET VARIABLE
called without arguments
ASTERISK-25662: Malformed AGI 520 Usage response
Reported by: Tony Mountifield
* [a007e438c3] Sean Bright -- res_agi: Fix malformed AGI usage response
Category: Resources/res_ari
ASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists
Reported by: Ronald Raikes
* [7901b9853e] George Joseph -- res_ari: Add "module loaded" check to
ari stubs
Category: Resources/res_ari_recordings
ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error
Reported by: Tim Morgan
* [cf6cf59646] Sean Bright -- stasis_recording: Correct ast_asprintf
error checking
Category: Resources/res_format_attr_h264
ASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional
parameters have a space
Reported by: John Harris
* [700ef6861a] Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp
attributes
Category: Resources/res_parking
ASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumasters
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
Category: Resources/res_pjsip
ASTERISK-27090: PJSIP: Deadlock using TCP transport
Reported by: Richard Mudgett
* [0d64cbde57] Richard Mudgett -- pjsip_distributor.c: Fix deadlock with
TCP type transports.
Category: Resources/res_pjsip/Bundling
ASTERISK-27052: Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow network
Reported by: alex
* [0bde568669] George Joseph -- pjproject_bundled: Use the asterisk
github mirror for download
Category: Resources/res_pjsip_refer
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin Harwell
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
during transfer
Category: Resources/res_pjsip_session
ASTERISK-27024: nat/external_media settings ignored in 14.4.1
Reported by: Christopher van de Sande
* [2dee95cc7a] Florian Floimair -- res_pjsip_session: Correct inverted
test in session_outgoing_nat_hook
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin Harwell
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
during transfer
ASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and
To URI differ
Reported by: Yasin CANER
* [36628cc9c4] Yasin CANER -- res_pjsip_session : fixed wrong From
Header number On Re-invite
Category: Resources/res_pjsip_transport_websocket
ASTERISK-27046: res_pjsip_transport_websocket: segfault in
get_write_timeout
Reported by: JA,rgen H
* [e16a669c70] JA,rgen H -- res_pjsip_transport_websocket: Add NULL
check in get_write_timeout
Category: Resources/res_rtp_asterisk
ASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component
Reported by: Michael Walton
* [7dafe82751] George Joseph -- res_rtp_asterisk: Fix ssrc change for
rtcp srtp
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
Reported by: Javier Riveros
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
wrong srtp unprotecting algorithm
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngstrAP:m
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
Category: Resources/res_srtp
ASTERISK-25294: srtp's crypto_get_random deprecated
Reported by: Tzafrir Cohen
* [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
Reported by: Javier Riveros
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
wrong srtp unprotecting algorithm
Category: Resources/res_stasis_snoop
ASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. Vaillancourt
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: pjproject/pjsip
ASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i)
Reported by: Matthias Binder
* [6af2dd34af] Alexei Gradinari -- res_pjsip: New endpoint option
"refer_blind_progress"
Improvement
Category: Core/BuildSystem
ASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL
Reported by: Guido Falsi
* [6a64f65fe6] Guido Falsi -- BuildSystem: Add patches to allow building
with recent LibreSSL
Category: Core/Channels
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
audiohooks
Reported by: Michael Walton
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Core/General
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
audiohooks
Reported by: Michael Walton
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Core/Portability
ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to
missing crypt.h file
Reported by: Guido Falsi
* [44cee2f4a1] Guido Falsi -- BuildSystem: Fix build on FreeBSD due to
missing crypt.h
Category: Resources/res_agi
ASTERISK-26124: res_agi: Set audio format for EAGI audio stream
Reported by: John Fawcett
* [90237dca11] Sean Bright -- res_agi: Allow configuration of audio
format of EAGI pipe
Category: Resources/res_pjsip_mwi
ASTERISK-26230: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP
taskprocessor on startup
Reported by: Alexei Gradinari
* [0f6a9617eb] Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI
subscriptions on updating contact
* [59c9bbe696] Alexei Gradinari -- res_pjsip_mwi: don't create mwi
subscriptions if initial unsolicited disabled
Category: Resources/res_rtp_asterisk
ASTERISK-26976: libsrtp-2.x.x support
Reported by: Alex
* [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
----------------------------------------------------------------------
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: Bridges/bridge_simple
ASTERISK-26469: Infinite loop after a dual Redirect
Reported by: Etienne Allovon
* [b07b216235] Joshua Colp -- manager: Clear the flag on the other
channel.
Category: Channels/chan_pjsip
ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite,
we're not trying UPDATE
Reported by: George Joseph
* [6bd7c0f37c] George Joseph -- chan_pjsip: Fix ability to send UPDATE
on COLP
Category: Core/Bridging
ASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events'
bridge is muted multiple times.
Reported by: Chris Howard
* [4910a3bf40] Joshua Colp -- channel: Fix reference counting in
ast_channel_suppress.
Category: General
ASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and
unidirectional negotiation
Reported by: Joshua Colp
* [0426b1d88a] Joshua Colp -- res_rtp_asterisk: Fix issues with ICE
renegotiation.
Category: Resources/res_corosync
ASTERISK-25370: res_corosync segfaults at startup with corosync version >
2.x
Reported by: mdu113
* [005a4afa6b] Jan Friesse -- res_corosync: Change thread stack size
Category: Resources/res_pjsip_dialog_info_body_generator
ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse
behavior difference between chan_sip and res_pjsip
Reported by: Zach R
* [a6e4899612] Alexei Gradinari -- res_pjsip: New endpoint option
"notify_early_inuse_ringing"
Category: Resources/res_pjsip_mwi
ASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on
deleting the endpoint's last contact
Reported by: Alexei Gradinari
* [8e749c8f51] Alexei Gradinari -- res_pjsip_mwi: unsubscribe
unsolicited MWI on deleting endpoint last contact
Category: Resources/res_stasis
ASTERISK-27059: res_stasis: Stolen channel references are leaking
Reported by: George Joseph
* [edfdb4dff5] George Joseph -- res_stasis: Plug reference leak on
stolen channels
Category: Third-Party/pjproject
ASTERISK-27097: pjproject_bundled: We don't pass options needed for
cross-compile to pjproject configure
Reported by: George Joseph
* [bbe68f139d] George Joseph -- pjproject_bundled: Allow passing
configure options to bundled
Improvement
Category: Applications/app_voicemail/IMAP
ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to
specify post-polling disconnect
Reported by: Alexei Gradinari
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
control
Category: Channels/chan_pjsip
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey Searle
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
Category: Resources/res_pjsip
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey Searle
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
----------------------------------------------------------------------
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 |
|------------+------------------+----------------------------------------|
| 0c00ee754b | George Joseph | Update for 13.17.0-rc1 |
|------------+------------------+----------------------------------------|
| 379fe65831 | George Joseph | Fix alembic branches |
|------------+------------------+----------------------------------------|
| 905d18e8bf | Richard Mudgett | pjsip_distributor.c: Fix |
| | | unidentified_requests hash functions. |
|------------+------------------+----------------------------------------|
| 1f59d08924 | Torrey Searle | res/res_pjsip_t38: fix incorrect |
| | | increment of media_count |
|------------+------------------+----------------------------------------|
| 764d04fa87 | Richard Mudgett | res_pjsip_mwi.c: Eliminate RAII_VAR in |
| | | contact delete observer |
|------------+------------------+----------------------------------------|
| cecf6540dc | Rodrigo RamArez | cdr: fix mistake spelling of a word |
| | Norambuena | for Unanswered. |
|------------+------------------+----------------------------------------|
| b9a4ab8c8c | Richard Mudgett | chan_pjsip: Fix PJSIP_MEDIA_OFFER |
| | | dialplan function read. |
|------------+------------------+----------------------------------------|
| f1a209d5ac | Richard Mudgett | app_voicemail.c: Fix compile error |
| | | when IMAP enabled. |
|------------+------------------+----------------------------------------|
| 68de35a6a0 | David M. Lee | CFLAGS for BIND8 support |
|------------+------------------+----------------------------------------|
| da3312457e | Sean Bright | codecs.conf.sample: Fix max_bandwidth |
| | | speling error |
|------------+------------------+----------------------------------------|
| 590ffcaf0b | Sean Bright | eventfd: Disable during cross |
| | | compilation |
|------------+------------------+----------------------------------------|
| 5520b6c201 | Alexei Gradinari | CHANGES: correct version for a new |
| | | option 'refer_blind_progress' |
|------------+------------------+----------------------------------------|
| c093bf8072 | Sean Bright | res_rtp_multicast: Use consistent |
| | | timestamps when possible |
|------------+------------------+----------------------------------------|
| c10341646d | George Joseph | test_json: Fix test names with |
| | | reserved words |
|------------+------------------+----------------------------------------|
| 65898c3af8 | George Joseph | unittests: Add a unit test that causes |
| | | a SEGV and... |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-13.16.0-summary.html | 405 ---
asterisk-13.16.0-summary.txt | 952 ---------
b/.version | 2
b/CHANGES | 54
b/ChangeLog | 1045 +++++++++-
b/Makefile | 3
b/addons/Makefile | 10
b/apps/app_chanspy.c | 16
b/apps/app_confbridge.c | 79
b/apps/app_dial.c | 6
b/apps/app_disa.c | 10
b/apps/app_dumpchan.c | 4
b/apps/app_externalivr.c | 6
b/apps/app_meetme.c | 2
b/apps/app_queue.c | 109 -
b/apps/app_voicemail.c | 80
b/asterisk-13.17.0-rc1-summary.html | 311 ++
b/asterisk-13.17.0-rc1-summary.txt | 832 +++++++
b/autoconf/ast_ext_lib.m4 | 36
b/bridges/bridge_native_rtp.c | 677 +++++-
b/bridges/bridge_simple.c | 32
b/channels/chan_pjsip.c | 68
b/channels/chan_sip.c | 8
b/channels/pjsip/dialplan_functions.c | 37
b/configs/samples/cdr.conf.sample | 2
b/configs/samples/codecs.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 20
b/configs/samples/sip.conf.sample | 3
b/configs/samples/voicemail.conf.sample | 3
b/configure | 434 +++-
b/configure.ac | 100
b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
b/contrib/realtime/mssql/mssql_config.sql | 46
b/contrib/realtime/mysql/mysql_config.sql | 18
b/contrib/realtime/oracle/oracle_config.sql | 46
b/contrib/realtime/postgresql/postgresql_config.sql | 22
b/formats/format_g729.c | 2
b/include/asterisk/ari.h | 10
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/bridge_channel.h | 2
b/include/asterisk/bridge_channel_internal.h | 11
b/include/asterisk/bridge_technology.h | 3
b/include/asterisk/channel.h | 25
b/include/asterisk/codec.h | 3
b/include/asterisk/core_local.h | 37
b/include/asterisk/format.h | 11
b/include/asterisk/res_pjsip.h | 74
b/include/asterisk/res_pjsip_presence_xml.h | 3
b/include/asterisk/res_pjsip_session.h | 11
b/include/asterisk/rtp_engine.h | 9
b/include/asterisk/smoother.h | 1
b/include/asterisk/test.h | 8
b/main/autoservice.c | 2
b/main/bridge.c | 10
b/main/bridge_after.c | 2
b/main/bridge_channel.c | 38
b/main/channel.c | 90
b/main/codec_builtin.c | 19
b/main/core_local.c | 54
b/main/crypt.c | 2
b/main/data.c | 4
b/main/file.c | 20
b/main/format.c | 8
b/main/libasteriskssl.c | 4
b/main/manager.c | 8
b/main/pbx.c | 4
b/main/pbx_app.c | 7
b/main/pbx_builtins.c | 8
b/main/tcptls.c | 4
b/main/test.c | 4
b/makeopts.in | 2
b/res/res_agi.c | 73
b/res/res_ari_applications.c | 4
b/res/res_ari_asterisk.c | 4
b/res/res_ari_bridges.c | 4
b/res/res_ari_channels.c | 4
b/res/res_ari_device_states.c | 4
b/res/res_ari_endpoints.c | 4
b/res/res_ari_events.c | 33
b/res/res_ari_mailboxes.c | 4
b/res/res_ari_playbacks.c | 4
b/res/res_ari_recordings.c | 4
b/res/res_ari_sounds.c | 4
b/res/res_corosync.c | 29
b/res/res_format_attr_h263.c | 2
b/res/res_format_attr_h264.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip.c | 31
b/res/res_pjsip/location.c | 53
b/res/res_pjsip/pjsip_configuration.c | 9
b/res/res_pjsip/pjsip_distributor.c | 242 +-
b/res/res_pjsip/presence_xml.c | 9
b/res/res_pjsip_dialog_info_body_generator.c | 10
b/res/res_pjsip_mwi.c | 87
b/res/res_pjsip_pidf_body_generator.c | 2
b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 8
b/res/res_pjsip_refer.c | 28
b/res/res_pjsip_sdp_rtp.c | 38
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_session.exports.in | 1
b/res/res_pjsip_t38.c | 2
b/res/res_pjsip_transport_websocket.c | 4
b/res/res_pjsip_xpidf_body_generator.c | 2
b/res/res_rtp_asterisk.c | 41
b/res/res_rtp_multicast.c | 139 +
b/res/res_srtp.c | 15
b/res/res_stasis.c | 20
b/res/srtp/srtp_compat.h | 29
b/res/stasis_recording/stored.c | 4
b/rest-api-templates/res_ari_resource.c.mustache | 35
b/tests/test_bridging.c | 292 ++
b/tests/test_json.c | 16
b/tests/test_pbx.c | 22
b/third-party/configure.m4 | 5
b/third-party/pjproject/Makefile | 2
b/third-party/pjproject/Makefile.rules | 7
b/third-party/pjproject/configure.m4 | 24
b/third-party/pjproject/patches/0070-Set-PJSIP_INV_SUPPORT_UPDATE-correctly-in-pjsip_inv_.patch | 16
121 files changed, 5477 insertions(+), 2043 deletions(-)

42
asterisk-13.17.1-summary.html Normal file
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@@ -0,0 +1,42 @@
<!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-13.17.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.17.1</h3><h3 align="center">Date: 2017-08-31</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 has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
<li><a href="http://downloads.asterisk.org/pub/security/AST-2017-005,AST-2017-006,AST-2017-007.html">AST-2017-005,AST-2017-006,AST-2017-007</a></li>
</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.17.0.</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%">1 George Joseph <gjoseph@digium.com><br/>1 Corey Farrell <git@cfware.com><br/>1 Joshua Colp <jcolp@digium.com><br/></td><td width="33%"><td width="33%">1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Corey Farrell <git@cfware.com><br/>1 Ross Beer<br/>1 Joshua Colp <jcolp@digium.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: Applications/app_minivm</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Applications/app_system</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Functions/func_shell</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27152">ASTERISK-27152</a>: Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0e5b7743d9c038d49c0fb10382fdbac21b212040">[0e5b7743d9]</a> George Joseph -- pjsip_message_ip_updater: Fix issue handling "tel" URIs</li>
</ul><br><h4>Category: Resources/res_monitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27103">ASTERISK-27103</a>: core: ast_safe_system command injection possible.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=707892089dccccb8dba2fe572cdeb759a449640a">[707892089d]</a> Corey Farrell -- AST-2017-006: Fix app_minivm application MinivmNotify command injection</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27013">ASTERISK-27013</a>: res_rtp_asterisk: Media can be hijacked even with strict RTP enabled<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3ee5c6dcbe6cec373f2519c70ab575955b532cd5">[3ee5c6dcbe]</a> Joshua Colp -- res_rtp_asterisk: Only learn a new source in learn state.</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>README-SERIOUSLY.bestpractices.txt | 7 ++
apps/app_minivm.c | 36 ++++++++----
apps/app_mixmonitor.c | 15 +++++
apps/app_system.c | 10 +++
configs/samples/minivm.conf.sample | 2
funcs/func_shell.c | 5 +
include/asterisk/app.h | 31 +++++++++-
main/asterisk.c | 91 ++++++++++++++++++++++++++-----
res/res_monitor.c | 13 +++-
res/res_pjsip/pjsip_message_ip_updater.c | 56 ++++++++++++++-----
res/res_rtp_asterisk.c | 79 ++++++++++++++------------
11 files changed, 265 insertions(+), 80 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.17.1
Date: 2017-08-31
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release has been made to address one or more security vulnerabilities
that have been identified. A security advisory document has been published
for each vulnerability that includes additional information. Users of
versions of Asterisk that are affected are strongly encouraged to review
the advisories and determine what action they should take to protect their
systems from these issues.
Security Advisories:
* AST-2017-005,AST-2017-006,AST-2017-007
The data in this summary reflects changes that have been made since the
previous release, asterisk-13.17.0.
----------------------------------------------------------------------
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
1 George Joseph 1 Ross Beer
1 Corey Farrell 1 Corey Farrell
1 Joshua Colp 1 Ross Beer
1 Joshua Colp
----------------------------------------------------------------------
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: Applications/app_minivm
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Applications/app_mixmonitor
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Applications/app_system
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Applications/app_voicemail
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Channels/chan_dahdi
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Core/General
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Functions/func_shell
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: General
ASTERISK-27152: Sending a "tel" uri in a From or To header in an
unauthenticated message causes asterisk to crash
Reported by: Ross Beer
* [0e5b7743d9] George Joseph -- pjsip_message_ip_updater: Fix issue
handling "tel" URIs
Category: Resources/res_monitor
ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell
* [707892089d] Corey Farrell -- AST-2017-006: Fix app_minivm application
MinivmNotify command injection
Category: Resources/res_rtp_asterisk
ASTERISK-27013: res_rtp_asterisk: Media can be hijacked even with strict
RTP enabled
Reported by: Joshua Colp
* [3ee5c6dcbe] Joshua Colp -- res_rtp_asterisk: Only learn a new source
in learn state.
----------------------------------------------------------------------
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.
README-SERIOUSLY.bestpractices.txt | 7 ++
apps/app_minivm.c | 36 ++++++++----
apps/app_mixmonitor.c | 15 +++++
apps/app_system.c | 10 +++
configs/samples/minivm.conf.sample | 2
funcs/func_shell.c | 5 +
include/asterisk/app.h | 31 +++++++++-
main/asterisk.c | 91 ++++++++++++++++++++++++++-----
res/res_monitor.c | 13 +++-
res/res_pjsip/pjsip_message_ip_updater.c | 56 ++++++++++++++-----
res/res_rtp_asterisk.c | 79 ++++++++++++++------------
11 files changed, 265 insertions(+), 80 deletions(-)