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Update for 16.16.1

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Asterisk Development Team
2021-02-18 11:48:09 -05:00
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16.16.0
16.16.1

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ChangeLog
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2021-02-18 16:48 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 16.16.1 Released.
2021-02-01 15:24 +0000 [a5619097cd] Kevin Harwell <kharwell@sangoma.com>
* AST-2021-002: Remote crash possible when negotiating T.38
When an endpoint requests to re-negotiate for fax and the incoming
re-invite is received prior to Asterisk sending out the 200 OK for
the initial invite the re-invite gets delayed. When Asterisk does
finally send the re-inivite the SDP includes streams for both audio
and T.38.
This happens because when the pending topology and active topologies
differ (pending stream is not in the active) in the delayed scenario
the pending stream is appended to the active topology. However, in
the fax case the pending stream should replace the active.
This patch makes it so when a delay occurs during fax negotiation,
to or from, the audio stream is replaced by the T.38 stream, or vice
versa instead of being appended.
Further when Asterisk sent the re-invite with both audio and T.38,
and the endpoint responded with a declined T.38 stream then Asterisk
would crash when attempting to change the T.38 state.
This patch also puts in a check that ensures the media state has a
valid fax session (associated udptl object) before changing the
T.38 state internally.
ASTERISK-29203 #close
Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09
2021-01-26 11:09 +0000 [3f4dfd5c02] Alexander Traud <pabstraud@compuserve.com>
* rtp: Enable srtp replay protection
Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.
ASTERISK-29260
Reported by: Alexander Traud
Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458
2020-12-28 06:43 +0000 [17561b5e64] Ivan Poddubnyi <ivan.poddubny@gmail.com>
* res_pjsip_diversion: Fix adding more than one histinfo to Supported
New responses sent within a PJSIP sessions are based on those that were
sent before. Therefore, adding/modifying a header once causes it to be
sent on all responses that follow.
Sending 181 Call Is Being Forwarded many times first adds "histinfo"
duplicated more and more, and eventually overflows past the array
boundary.
This commit adds a check preventing adding "histinfo" more than once,
and skipping it if there is no more space in the header.
Similar overflow situations can also occur in res_pjsip_path and
res_pjsip_outbound_registration so those were also modified to
check the bounds and suppress duplicate Supported values.
ASTERISK-29227
Reported by: Ivan Poddubny
Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322
2020-12-11 14:49 +0000 [4cea145aa9] Sean Bright <sean.bright@gmail.com>
* res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
ASTERISK-29205 #close
Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea
2021-02-05 05:26 +0000 [321632b02e] Joshua C. Colp <jcolp@sangoma.com>
* pjsip: Make modify_local_offer2 tolerate previous failed SDP.
If a remote side is broken and sends an SDP that can not be
negotiated the call will be torn down but there is a window
where a second 183 Session Progress or 200 OK that is forked
can be received that also attempts to negotiate SDP. Since
the code marked the SDP negotiation as being done and complete
prior to this it assumes that there is an active local and remote
SDP which it can modify, while in fact there is not as the SDP
did not successfully negotiate. Since there is no local or remote
SDP a crash occurs.
This patch changes the pjmedia_sdp_neg_modify_local_offer2
function to no longer assume that a previous SDP negotiation
was successful.
ASTERISK-29196
Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e
2021-01-21 16:28 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 16.16.0 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-16.16.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.16.0</h3><h3 align="center">Date: 2021-01-21</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#open_issues">Open Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.15.0.</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%">6 Sean Bright <sean.bright@gmail.com><br/>4 Alexander Traud <pabstraud@compuserve.com><br/>3 George Joseph <gjoseph@digium.com><br/>3 Jaco Kroon <jaco@uls.co.za><br/>3 Joshua C. Colp <jcolp@sangoma.com><br/>2 Asterisk Development Team <asteriskteam@digium.com><br/>2 Ivan Poddubnyi <ivan.poddubny@gmail.com><br/>2 Sungtae Kim <pchero21@gmail.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Kevin Harwell <kharwell@sangoma.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Jean Aunis <jean.aunis@prescom.fr><br/>1 Torrey Searle <tsearle@voxbone.com><br/>1 laszlovl <digium@lvlconsultancy.nl><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Nathan Bruning <nathan@iperity.com><br/>1 Pirmin Walthert <infos@nappsoft.ch><br/>1 Stanislav <stas.abramenkov@gmail.com><br/>1 Alexander Greiner-Baer <alex+asterisk@greiner-baer.de><br/></td><td width="33%">1 Mark Petersen<br/></td><td width="33%">4 Alexander Traud <pabstraud@compuserve.com><br/>2 Sean Bright <sean.bright@gmail.com><br/>2 sungtae kim <pchero21@gmail.com><br/>2 George Joseph <gjoseph@digium.com><br/>1 Flole Systems <flole@flole.de><br/>1 Michael Maier<br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Julien <tigood@gmail.com><br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Hendrik Wedhorn <hwedhorn@addix.net><br/>1 Robert Sutton <rsutton@noojee.com.au><br/>1 Alex Hermann<br/>1 Alex Hermann <alex-asterisk@hexla.nl><br/>1 Juan Carlos Castro y Castro <jccyc1965@gmail.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Alexander Greiner-Baer <alex+asterisk@greiner-baer.de><br/>1 Alexander Traud<br/>1 Mark Petersen <bugs.digium.com@zombie.dk><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Nathan Bruning <nathan@iperity.com><br/>1 Mark Petersen<br/>1 Michael Maier <m1278468@mailbox.org><br/>1 Gant Liu <tpzzs@163.com><br/>1 Schneur Rosenberg <thesipguy@gmail.com><br/>1 Dan Cropp<br/>1 Stanislav Abramenkov <stas.abramenkov@gmail.com><br/>1 Torrey Searle <tsearle@gmail.com><br/>1 laszlovl <digium@lvlconsultancy.nl><br/>1 Mikhail Ivanov <mivanov@lanta-net.ru><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29219">ASTERISK-29219</a>: res_pjsip_diversion: Crash if Tel URI contains History-Info<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9196e0d1d5fd1f5e508bc4de90ac27d55b1b336a">[9196e0d1d5]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
</ul><br><h3>Bug</h3><h4>Category: Applications/app_chanspy</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28883">ASTERISK-28883</a>: Spyee information ist missing in ChanSpyStop AMI Event<br/>Reported by: Hendrik Wedhorn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a23296834c6f238bf1a3baa1a39606528b35ac2">[0a23296834]</a> Sean Bright -- app_chanspy: Spyee information missing in ChanSpyStop AMI Event</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28947">ASTERISK-28947</a>: Segmentation fault in mixmonitor_ds_destroy<br/>Reported by: Robert Sutton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e96f744816bb7a3847c5712d6f90148f49db04d9">[e96f744816]</a> Kevin Harwell -- app_mixmonitor: cleanup datastore when monitor thread fails to launch</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29155">ASTERISK-29155</a>: app_queue: Deadlock between queues container and individual queues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d8c9db6187216215cd516e76151583509c23470">[8d8c9db618]</a> George Joseph -- app_queue: Fix deadlock between update and show queues</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29161">ASTERISK-29161</a>: Incorrect setup of recall channels<br/>Reported by: Boris P. Korzun<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89d3de37ca3919f962130598c79c0492f9d312f7">[89d3de37ca]</a> Boris P. Korzun -- bridge_basic: Fixed setup of recall channels</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29240">ASTERISK-29240</a>: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable<br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97afc9055fbe8ca573a1a3745b7f4facff364d39">[97afc9055f]</a> Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after creating a channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27902">ASTERISK-27902</a>: chan_pjsip isn't updating hangupcause on 4XX responses<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17fa9c93d024444aabdc192b7be145b4b7e9380d">[17fa9c93d0]</a> Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28016">ASTERISK-28016</a>: PJSIP sends duplicate 183 Progress responses<br/>Reported by: Alex Hermann<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17fa9c93d024444aabdc192b7be145b4b7e9380d">[17fa9c93d0]</a> Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28185">ASTERISK-28185</a>: chan_pjsip: Subsequent same responses are not stopped<br/>Reported by: Julien<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17fa9c93d024444aabdc192b7be145b4b7e9380d">[17fa9c93d0]</a> Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29230">ASTERISK-29230</a>: pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send<br/>Reported by: Michael Maier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ed20b9d3b8447f54316af01c523231bf7dafffc">[7ed20b9d3b]</a> George Joseph -- Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff"</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29201">ASTERISK-29201</a>: Crash occurs when Transfer and execute Hangup before the Transfer result <br/>Reported by: Dan Cropp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e127a5776154d3f1abed916c0379d7a8610ccc7e">[e127a57761]</a> Dan Cropp -- chan_pjsip: Incorporate channel reference count into transfer_refer().</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29022">ASTERISK-29022</a>: Crash when manipulating PJSIP invite dlg ref counts<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea744ca7c25c1fe5cd9ac860bf54805edd741a1f">[ea744ca7c2]</a> Joshua C. Colp -- pjsip: Match lifetime of INVITE session to our session.</li>
</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29238">ASTERISK-29238</a>: chan_sip: SDP: Offers without any enabled stream are accepted.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b0960310011f9792c78c6e58ce2a8e49b345a6">[34b0960310]</a> Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29237">ASTERISK-29237</a>: chan_sip: SDP: m=video is parsed even when disabled.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b0960310011f9792c78c6e58ce2a8e49b345a6">[34b0960310]</a> Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.</li>
</ul><br><h4>Category: Channels/chan_sip/SRTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29222">ASTERISK-29222</a>: chan_sip: Hold/Resume an sRTP call on a video enabled user-agent.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b0960310011f9792c78c6e58ce2a8e49b345a6">[34b0960310]</a> Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.</li>
</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28798">ASTERISK-28798</a>: [patch] chan_sip: TCP/TLS client without server.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f667c5a781a75ef8f9f7b34875d1a639b55494c2">[f667c5a781]</a> Alexander Traud -- chan_sip: Remove unused sip_socket->port.</li>
</ul><br><h4>Category: Channels/chan_sip/Video</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29238">ASTERISK-29238</a>: chan_sip: SDP: Offers without any enabled stream are accepted.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b0960310011f9792c78c6e58ce2a8e49b345a6">[34b0960310]</a> Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29237">ASTERISK-29237</a>: chan_sip: SDP: m=video is parsed even when disabled.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b0960310011f9792c78c6e58ce2a8e49b345a6">[34b0960310]</a> Alexander Traud -- chan_sip: SDP: Sidestep stream parsing when its media is disabled.</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29209">ASTERISK-29209</a>: Debug messages printed by scope trace might be missing newlines<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a2867efa9166f2fd8e327daa9ce0b794eca3e2d">[5a2867efa9]</a> George Joseph -- logger.c: Automatically add a newline to formats that don't have one</li>
</ul><br><h4>Category: Functions/func_lock</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29217">ASTERISK-29217</a>: LOCK() can grant the same lock to multiple channels spuriously<br/>Reported by: Jaco Kroon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32e36144c74ddbcae26f633026eeee6f190911a2">[32e36144c7]</a> Jaco Kroon -- func_lock: fix multiple-channel-grant problems.</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29148">ASTERISK-29148</a>: AST_MODULE_INFO no, MODULEINFO depend<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c79bc19d1f85415372ce992b474ea015257f279">[4c79bc19d1]</a> Alexander Traud -- loader: Sync load- and build-time deps.</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29188">ASTERISK-29188</a>: null media causing the Asterisk crash<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a47e6965b3b304b313304a0df56753b137dd6fd3">[a47e6965b3]</a> Sungtae Kim -- res_ari: Fix wrong media uri handle for channel play</li>
</ul><br><h4>Category: Resources/res_http_media_cache</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29173">ASTERISK-29173</a>: Media cache URL requests allow infinite redirects<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c185c9e21f7f0bd9c9f7a1a1cda8f33e12c2a2a">[0c185c9e21]</a> Sean Bright -- res_http_media_cache.c: Set reasonable number of redirects</li>
</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29211">ASTERISK-29211</a>: res_musiconhold: Segfault on realtime music on hold without entries<br/>Reported by: Nathan Bruning<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb46595799c65c10df18730ed93d4c16bcc0e503">[bb46595799]</a> Nathan Bruning -- res_musiconhold: Don't crash when real-time doesn't return any entries</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29165">ASTERISK-29165</a>: res_pjsip: malformed header Accept-Encoding in OPTIONS response<br/>Reported by: Alexander Greiner-Baer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a8f6238cc8cbb9a19bc5e981c6d08aa4aa020a88">[a8f6238cc8]</a> Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding to identity in OPTIONS response</li>
</ul><br><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9196e0d1d5fd1f5e508bc4de90ac27d55b1b336a">[9196e0d1d5]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29231">ASTERISK-29231</a>: pjsip: SIGSEGV in CLI if no trunk is registered<br/>Reported by: Michael Maier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ed20b9d3b8447f54316af01c523231bf7dafffc">[7ed20b9d3b]</a> George Joseph -- Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff"</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29240">ASTERISK-29240</a>: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable<br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97afc9055fbe8ca573a1a3745b7f4facff364d39">[97afc9055f]</a> Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after creating a channel</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29229">ASTERISK-29229</a>: Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription<br/>Reported by: Jean Aunis - Prescom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45e1d89135a636e1177c3fd767846a066d45bbfa">[45e1d89135]</a> Jean Aunis -- Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.</li>
</ul><br><h4>Category: Resources/res_stir_shaken</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29175">ASTERISK-29175</a>: res_pjsip_stir_shaken: Fix module description<br/>Reported by: Stanislav Abramenkov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=159522003ab1270140d312bb5a823a3a6502d1a6">[159522003a]</a> Stanislav -- res_pjsip_stir_shaken: Fix module description</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9196e0d1d5fd1f5e508bc4de90ac27d55b1b336a">[9196e0d1d5]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29024">ASTERISK-29024</a>: pjsip: Route Header in Cancel request incorrectly set<br/>Reported by: Flole Systems<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=11def974a84ea523b90e0ec6e9af43ff58b6eb14">[11def974a8]</a> Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of strings when appropriate</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29118">ASTERISK-29118</a>: VoiceMail() should have an option to play greetings as Early Media<br/>Reported by: Juan Carlos Castro y Castro<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=15566494f968adafbe5b89555c7abc63a1e36cd1">[15566494f9]</a> Joshua C. Colp -- voicemail: add option 'e' to play greetings as early media</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28549">ASTERISK-28549</a>: Two repeated 183<br/>Reported by: Gant Liu<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17fa9c93d024444aabdc192b7be145b4b7e9380d">[17fa9c93d0]</a> Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29216">ASTERISK-29216</a>: contrib: systemd asterisk service for centos8 or other newer linux versions<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7df88c98d034f8b918e4ff86c9d0756f54aad0c4">[7df88c98d0]</a> Jaco Kroon -- contrib/systemd: Added note on common issues with systemd and asterisk</li>
</ul><br><h4>Category: Resources/res_http_media_cache</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29143">ASTERISK-29143</a>: res_http_media_cache: HTTP media cache stored hardcoded in /tmp<br/>Reported by: laszlovl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d2558209ba8c7fae77d740dc8dd377d5beb8b8e">[8d2558209b]</a> laszlovl -- Introduce astcachedir, to be used for temporary bucket files</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28549">ASTERISK-28549</a>: Two repeated 183<br/>Reported by: Gant Liu<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17fa9c93d024444aabdc192b7be145b4b7e9380d">[17fa9c93d0]</a> Ivan Poddubnyi -- chan_pjsip: Stop queueing control frames twice on outgoing channels</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: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28992">ASTERISK-28992</a>: app_voicemail: Deadlock in ODBC when retrieving file<br/>Reported by: Schneur Rosenberg<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b7af3eb27d07d3de86943f9025df343ee447e08">[2b7af3eb27]</a> Sean Bright -- app_voicemail: Prevent deadlocks when out of ODBC database connections</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29215">ASTERISK-29215</a>: res_pjsip_session: NULL active_media_state topology caused asterisk crash<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab3f57d88f2ba8166c3684c10956caade3966e2b">[ab3f57d88f]</a> Sungtae Kim -- res_pjsip_session: Fixed NULL active media topology handle</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=e08d866884b5e9195520e12209eb86df140415cd">e08d866884</a></td><td>Asterisk Development Team</td><td>Update for 16.16.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6056818467833d24fbd6491688306f0bb9bc1ba3">6056818467</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.16.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aca435dfe7daed36a05ec59b78a27d7372ea6320">aca435dfe7</a></td><td>Jaco Kroon</td><td>pbx_lua: Add LUA_VERSIONS environment variable to ./configure.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c5bffb217519cfa0e8fe9b430d80e612da8b224">4c5bffb217</a></td><td>Sean Bright</td><td>asterisk: Export additional manager functions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89cf7899be5d0e278fb5a406cf3e2484e064db6f">89cf7899be</a></td><td>Richard Mudgett</td><td>res_pjsip_session.c: Fix compiler warnings.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7e4bb4ed11b2741ff6cd47a95fb6e815a5e1d901">7e4bb4ed11</a></td><td>Joshua C. Colp</td><td>res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddbf3a7f73041c2de39f5e8f1518240507e38b24">ddbf3a7f73</a></td><td>Sean Bright</td><td>media_cache: Fix reference leak with bucket file metadata</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a360150ee025108d8e150423df38878132fe71dd">a360150ee0</a></td><td>Sean Bright</td><td>CHANGES: Remove already applied CHANGES update</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1a78e047d1f4e80a8bb7c97adce3f19713a1ca2">d1a78e047d</a></td><td>Alexander Traud</td><td>modules.conf: Align the comments for more conclusiveness.</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-16.15.0-summary.html | 213 ----------
asterisk-16.15.0-summary.txt | 543 --------------------------
b/.version | 2
b/CHANGES | 18
b/ChangeLog | 507 ++++++++++++++++++++++++
b/Makefile | 6
b/apps/app_chanspy.c | 6
b/apps/app_mixmonitor.c | 23 +
b/apps/app_queue.c | 245 ++++++-----
b/apps/app_voicemail.c | 36 +
b/asterisk-16.16.0-rc1-summary.html | 164 +++++++
b/asterisk-16.16.0-rc1-summary.txt | 492 +++++++++++++++++++++++
b/build_tools/install_subst | 1
b/build_tools/make_defaults_h | 1
b/build_tools/mkpkgconfig | 1
b/channels/chan_pjsip.c | 214 +++-------
b/channels/chan_sip.c | 32 -
b/channels/sip/include/sip.h | 2
b/configs/basic-pbx/modules.conf | 8
b/configs/samples/asterisk.conf.sample | 1
b/configs/samples/modules.conf.sample | 9
b/configure | 11
b/configure.ac | 9
b/contrib/systemd/asterisk.service | 7
b/funcs/func_lock.c | 163 ++-----
b/funcs/func_odbc.c | 1
b/funcs/func_periodic_hook.c | 1
b/include/asterisk/manager.h | 4
b/include/asterisk/paths.h | 1
b/main/asterisk.c | 4
b/main/bridge_basic.c | 2
b/main/bucket.c | 3
b/main/logger.c | 5
b/main/manager.c | 6
b/main/manager_channels.c | 18
b/main/media_cache.c | 1
b/main/options.c | 7
b/main/pbx_variables.c | 2
b/makeopts.in | 1
b/res/res_hep_pjsip.c | 2
b/res/res_http_media_cache.c | 1
b/res/res_musiconhold.c | 21 -
b/res/res_odbc.c | 1
b/res/res_pjproject.c | 2
b/res/res_pjsip.c | 2
b/res/res_pjsip/pjsip_options.c | 2
b/res/res_pjsip_diversion.c | 11
b/res/res_pjsip_dlg_options.c | 2
b/res/res_pjsip_nat.c | 10
b/res/res_pjsip_outbound_registration.c | 296 +++++---------
b/res/res_pjsip_pidf_digium_body_supplement.c | 8
b/res/res_pjsip_session.c | 66 +--
b/res/res_pjsip_stir_shaken.c | 4
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_stasis_playback.c | 7
b/res/res_stasis_snoop.c | 12
b/res/stasis/messaging.c | 33 +
doc/CHANGES-staging/hide_messaging_ami_events | 11
58 files changed, 1826 insertions(+), 1437 deletions(-)</pre><br></html>

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@@ -1,486 +0,0 @@
Release Summary
asterisk-16.16.0
Date: 2021-01-21
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Open Issues
5. Other Changes
6. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-16.15.0.
----------------------------------------------------------------------
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
6 Sean Bright 1 Mark Petersen 4 Alexander Traud
4 Alexander Traud 2 Sean Bright
3 George Joseph 2 sungtae kim
3 Jaco Kroon 2 George Joseph
3 Joshua C. Colp 1 Flole Systems
2 Asterisk Development Team 1 Michael Maier
2 Ivan Poddubnyi 1 Ivan Poddubny
2 Sungtae Kim 1 Julien
1 Dan Cropp 1 Jaco Kroon
1 Kevin Harwell 1 Jean Aunis - Prescom
1 Boris P. Korzun 1 Hendrik Wedhorn
1 Jean Aunis 1 Robert Sutton
1 Torrey Searle 1 Alex Hermann
1 laszlovl 1 Alex Hermann
1 Richard Mudgett 1 Juan Carlos Castro y Castro
1 Nathan Bruning 1 Boris P. Korzun
1 Pirmin Walthert 1 Alexander Greiner-Baer
1 Stanislav 1 Alexander Traud
1 Alexander Greiner-Baer 1 Mark Petersen
1 Dan Cropp
1 Nathan Bruning
1 Mark Petersen
1 Michael Maier
1 Gant Liu
1 Schneur Rosenberg
1 Dan Cropp
1 Stanislav Abramenkov
1 Torrey Searle
1 laszlovl
1 Mikhail Ivanov
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Security
Category: Resources/res_pjsip_diversion
ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains
History-Info
Reported by: Torrey Searle
* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
Bug
Category: Applications/app_chanspy
ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
Reported by: Hendrik Wedhorn
* [0a23296834] Sean Bright -- app_chanspy: Spyee information missing in
ChanSpyStop AMI Event
Category: Applications/app_mixmonitor
ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
Reported by: Robert Sutton
* [e96f744816] Kevin Harwell -- app_mixmonitor: cleanup datastore when
monitor thread fails to launch
Category: Applications/app_queue
ASTERISK-29155: app_queue: Deadlock between queues container and
individual queues
Reported by: George Joseph
* [8d8c9db618] George Joseph -- app_queue: Fix deadlock between update
and show queues
Category: Bridges/bridge_simple
ASTERISK-29161: Incorrect setup of recall channels
Reported by: Boris P. Korzun
* [89d3de37ca] Boris P. Korzun -- bridge_basic: Fixed setup of recall
channels
Category: Channels/chan_pjsip
ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
instead of a channel variable
Reported by: Ivan Poddubny
* [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
creating a channel
ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses
Reported by: George Joseph
* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
Reported by: Alex Hermann
* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
Reported by: Julien
* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if
registration can't be send
Reported by: Michael Maier
* [7ed20b9d3b] George Joseph -- Revert
"res_pjsip_outbound_registration.c: Use our own scheduler and other
stuff"
ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the
Transfer result
Reported by: Dan Cropp
* [e127a57761] Dan Cropp -- chan_pjsip: Incorporate channel reference
count into transfer_refer().
ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts
Reported by: Sean Bright
* [ea744ca7c2] Joshua C. Colp -- pjsip: Match lifetime of INVITE session
to our session.
Category: Channels/chan_sip/CodecHandling
ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
accepted.
Reported by: Alexander Traud
* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Channels/chan_sip/SRTP
ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled
user-agent.
Reported by: Alexander Traud
* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Channels/chan_sip/TCP-TLS
ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
Reported by: Alexander Traud
* [f667c5a781] Alexander Traud -- chan_sip: Remove unused
sip_socket->port.
Category: Channels/chan_sip/Video
ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
accepted.
Reported by: Alexander Traud
* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
* [34b0960310] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Core/Logging
ASTERISK-29209: Debug messages printed by scope trace might be missing
newlines
Reported by: Alexander Traud
* [5a2867efa9] George Joseph -- logger.c: Automatically add a newline to
formats that don't have one
Category: Functions/func_lock
ASTERISK-29217: LOCK() can grant the same lock to multiple channels
spuriously
Reported by: Jaco Kroon
* [32e36144c7] Jaco Kroon -- func_lock: fix multiple-channel-grant
problems.
Category: General
ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend
Reported by: Alexander Traud
* [4c79bc19d1] Alexander Traud -- loader: Sync load- and build-time
deps.
Category: Resources/res_ari_channels
ASTERISK-29188: null media causing the Asterisk crash
Reported by: sungtae kim
* [a47e6965b3] Sungtae Kim -- res_ari: Fix wrong media uri handle for
channel play
Category: Resources/res_http_media_cache
ASTERISK-29173: Media cache URL requests allow infinite redirects
Reported by: Sean Bright
* [0c185c9e21] Sean Bright -- res_http_media_cache.c: Set reasonable
number of redirects
Category: Resources/res_musiconhold
ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold
without entries
Reported by: Nathan Bruning
* [bb46595799] Nathan Bruning -- res_musiconhold: Don't crash when
real-time doesn't return any entries
Category: Resources/res_pjsip
ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS
response
Reported by: Alexander Greiner-Baer
* [a8f6238cc8] Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding
to identity in OPTIONS response
Category: Resources/res_pjsip_diversion
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
Category: Resources/res_pjsip_outbound_registration
ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered
Reported by: Michael Maier
* [7ed20b9d3b] George Joseph -- Revert
"res_pjsip_outbound_registration.c: Use our own scheduler and other
stuff"
Category: Resources/res_pjsip_session
ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
instead of a channel variable
Reported by: Ivan Poddubny
* [97afc9055f] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
creating a channel
Category: Resources/res_stasis
ASTERISK-29229: Stasis/messaging: text messages not dispatched to all
subscribers when using generic subscription
Reported by: Jean Aunis - Prescom
* [45e1d89135] Jean Aunis -- Stasis/messaging: tech subscriptions
conflict with endpoint subscriptions.
Category: Resources/res_stir_shaken
ASTERISK-29175: res_pjsip_stir_shaken: Fix module description
Reported by: Stanislav Abramenkov
* [159522003a] Stanislav -- res_pjsip_stir_shaken: Fix module
description
Category: pjproject/pjsip
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [9196e0d1d5] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set
Reported by: Flole Systems
* [11def974a8] Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of
strings when appropriate
Improvement
Category: Applications/app_voicemail/NewFeature
ASTERISK-29118: VoiceMail() should have an option to play greetings as
Early Media
Reported by: Juan Carlos Castro y Castro
* [15566494f9] Joshua C. Colp -- voicemail: add option 'e' to play
greetings as early media
Category: Channels/chan_pjsip
ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
Category: Contrib/General
ASTERISK-29216: contrib: systemd asterisk service for centos8 or other
newer linux versions
Reported by: Mark Petersen
* [7df88c98d0] Jaco Kroon -- contrib/systemd: Added note on common
issues with systemd and asterisk
Category: Resources/res_http_media_cache
ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in
/tmp
Reported by: laszlovl
* [8d2558209b] laszlovl -- Introduce astcachedir, to be used for
temporary bucket files
Category: Resources/res_pjsip_session
ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
* [17fa9c93d0] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
----------------------------------------------------------------------
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: Applications/app_voicemail/ODBC
ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
Reported by: Schneur Rosenberg
* [2b7af3eb27] Sean Bright -- app_voicemail: Prevent deadlocks when out
of ODBC database connections
Category: Resources/res_pjsip_session
ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
asterisk crash
Reported by: sungtae kim
* [ab3f57d88f] Sungtae Kim -- res_pjsip_session: Fixed NULL active media
topology handle
----------------------------------------------------------------------
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 |
|------------+-------------+---------------------------------------------|
| | Asterisk | |
| e08d866884 | Development | Update for 16.16.0-rc1 |
| | Team | |
|------------+-------------+---------------------------------------------|
| | Asterisk | |
| 6056818467 | Development | Update CHANGES and UPGRADE.txt for 16.16.0 |
| | Team | |
|------------+-------------+---------------------------------------------|
| aca435dfe7 | Jaco Kroon | pbx_lua: Add LUA_VERSIONS environment |
| | | variable to ./configure. |
|------------+-------------+---------------------------------------------|
| 4c5bffb217 | Sean Bright | asterisk: Export additional manager |
| | | functions |
|------------+-------------+---------------------------------------------|
| 89cf7899be | Richard | res_pjsip_session.c: Fix compiler warnings. |
| | Mudgett | |
|------------+-------------+---------------------------------------------|
| 7e4bb4ed11 | Joshua C. | res_pjsip_pidf_digium_body_supplement: |
| | Colp | Support Sangoma user agent. |
|------------+-------------+---------------------------------------------|
| ddbf3a7f73 | Sean Bright | media_cache: Fix reference leak with bucket |
| | | file metadata |
|------------+-------------+---------------------------------------------|
| a360150ee0 | Sean Bright | CHANGES: Remove already applied CHANGES |
| | | update |
|------------+-------------+---------------------------------------------|
| d1a78e047d | Alexander | modules.conf: Align the comments for more |
| | Traud | conclusiveness. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-16.15.0-summary.html | 213 ----------
asterisk-16.15.0-summary.txt | 543 --------------------------
b/.version | 2
b/CHANGES | 18
b/ChangeLog | 507 ++++++++++++++++++++++++
b/Makefile | 6
b/apps/app_chanspy.c | 6
b/apps/app_mixmonitor.c | 23 +
b/apps/app_queue.c | 245 ++++++-----
b/apps/app_voicemail.c | 36 +
b/asterisk-16.16.0-rc1-summary.html | 164 +++++++
b/asterisk-16.16.0-rc1-summary.txt | 492 +++++++++++++++++++++++
b/build_tools/install_subst | 1
b/build_tools/make_defaults_h | 1
b/build_tools/mkpkgconfig | 1
b/channels/chan_pjsip.c | 214 +++-------
b/channels/chan_sip.c | 32 -
b/channels/sip/include/sip.h | 2
b/configs/basic-pbx/modules.conf | 8
b/configs/samples/asterisk.conf.sample | 1
b/configs/samples/modules.conf.sample | 9
b/configure | 11
b/configure.ac | 9
b/contrib/systemd/asterisk.service | 7
b/funcs/func_lock.c | 163 ++-----
b/funcs/func_odbc.c | 1
b/funcs/func_periodic_hook.c | 1
b/include/asterisk/manager.h | 4
b/include/asterisk/paths.h | 1
b/main/asterisk.c | 4
b/main/bridge_basic.c | 2
b/main/bucket.c | 3
b/main/logger.c | 5
b/main/manager.c | 6
b/main/manager_channels.c | 18
b/main/media_cache.c | 1
b/main/options.c | 7
b/main/pbx_variables.c | 2
b/makeopts.in | 1
b/res/res_hep_pjsip.c | 2
b/res/res_http_media_cache.c | 1
b/res/res_musiconhold.c | 21 -
b/res/res_odbc.c | 1
b/res/res_pjproject.c | 2
b/res/res_pjsip.c | 2
b/res/res_pjsip/pjsip_options.c | 2
b/res/res_pjsip_diversion.c | 11
b/res/res_pjsip_dlg_options.c | 2
b/res/res_pjsip_nat.c | 10
b/res/res_pjsip_outbound_registration.c | 296 +++++---------
b/res/res_pjsip_pidf_digium_body_supplement.c | 8
b/res/res_pjsip_session.c | 66 +--
b/res/res_pjsip_stir_shaken.c | 4
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_stasis_playback.c | 7
b/res/res_stasis_snoop.c | 12
b/res/stasis/messaging.c | 33 +
doc/CHANGES-staging/hide_messaging_ami_events | 11
58 files changed, 1826 insertions(+), 1437 deletions(-)

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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-16.16.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.16.1</h3><h3 align="center">Date: 2021-02-18</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-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005.html">AST-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005</a></li>
</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.16.0.</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%">1 Ivan Poddubnyi <ivan.poddubny@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Kevin Harwell <kharwell@sangoma.com><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Joshua C. Colp <jcolp@sangoma.com><br/></td><td width="33%"><td width="33%">1 Mauri de Souza Meneguzzo (3CPlus) <mauri.nunes@fluxoti.com><br/>1 Ivan Poddubny<br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Edvin Vidmar <edvinvidmar@hotmail.com><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Alexander Traud<br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29260">ASTERISK-29260</a>: sRTP Replay Protection ignored; even tears down long calls<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f4dfd5c022c82265cbaba7fa726d1e27dd21cce">[3f4dfd5c02]</a> Alexander Traud -- rtp: Enable srtp replay protection</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29227">ASTERISK-29227</a>: res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash<br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17561b5e643092b64a1e8351b25017b28ed690b4">[17561b5e64]</a> Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more than one histinfo to Supported</li>
</ul><br><h3>Bug</h3><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29196">ASTERISK-29196</a>: res_pjsip: Segmentation fault<br/>Reported by: Mauri de Souza Meneguzzo (3CPlus)<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=321632b02e828406d45bc4be2793399f1746c32a">[321632b02e]</a> Joshua C. Colp -- pjsip: Make modify_local_offer2 tolerate previous failed SDP.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5619097cd7e4da4d64a513d9659871ea217a706">[a5619097cd]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5619097cd7e4da4d64a513d9659871ea217a706">[a5619097cd]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29205">ASTERISK-29205</a>: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client<br/>Reported by: Edvin Vidmar<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cea145aa9a7dc90eabdd1f63cb3162f3e1d91e8">[4cea145aa9]</a> Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch that leads to overflow</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>configs/samples/rtp.conf.sample | 12 +++++++
doc/CHANGES-staging/srtp_replay_protection.txt | 9 +++++
doc/UPGRADE-staging/srtp_replay_protection.txt | 9 +++++
res/res_pjsip_diversion.c | 14 ++++++++
res/res_pjsip_outbound_registration.c | 12 +++++++
res/res_pjsip_path.c | 12 +++++++
res/res_pjsip_session.c | 9 +++++
res/res_pjsip_t38.c | 9 +++++
res/res_rtp_asterisk.c | 16 +++++++---
res/res_srtp.c | 5 +--
third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 15 +++++++++
11 files changed, 115 insertions(+), 7 deletions(-)</pre><br></html>

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Release Summary
asterisk-16.16.1
Date: 2021-02-18
<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-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005
The data in this summary reflects changes that have been made since the
previous release, asterisk-16.16.0.
----------------------------------------------------------------------
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
1 Ivan Poddubnyi 1 Mauri de Souza Meneguzzo (3CPlus)
1 Sean Bright 1 Ivan Poddubny
1 Kevin Harwell 1 Ivan Poddubny
1 Alexander Traud 1 Edvin Vidmar
1 Joshua C. Colp 1 Alexander Traud
1 Gregory Massel
1 Alexander Traud
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Security
Category: Resources/res_srtp
ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
Reported by: Alexander Traud
* [3f4dfd5c02] Alexander Traud -- rtp: Enable srtp replay protection
Category: pjproject/pjsip
ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes
memory corruption and crash
Reported by: Ivan Poddubny
* [17561b5e64] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more
than one histinfo to Supported
Bug
Category: Resources/res_pjsip
ASTERISK-29196: res_pjsip: Segmentation fault
Reported by: Mauri de Souza Meneguzzo (3CPlus)
* [321632b02e] Joshua C. Colp -- pjsip: Make modify_local_offer2
tolerate previous failed SDP.
Category: Resources/res_pjsip_session
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
Category: Resources/res_pjsip_t38
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
Category: Resources/res_rtp_asterisk
ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold
from webrtc client
Reported by: Edvin Vidmar
* [4cea145aa9] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch
that leads to overflow
----------------------------------------------------------------------
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.
configs/samples/rtp.conf.sample | 12 +++++++
doc/CHANGES-staging/srtp_replay_protection.txt | 9 +++++
doc/UPGRADE-staging/srtp_replay_protection.txt | 9 +++++
res/res_pjsip_diversion.c | 14 ++++++++
res/res_pjsip_outbound_registration.c | 12 +++++++
res/res_pjsip_path.c | 12 +++++++
res/res_pjsip_session.c | 9 +++++
res/res_pjsip_t38.c | 9 +++++
res/res_rtp_asterisk.c | 16 +++++++---
res/res_srtp.c | 5 +--
third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 15 +++++++++
11 files changed, 115 insertions(+), 7 deletions(-)