-16.16.0
\ No newline at end of file
+16.16.1
\ No newline at end of file
+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.
+++ /dev/null
-<!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"><asteriskteam@digium.com></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>
\ No newline at end of file
+++ /dev/null
- 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(-)
--- /dev/null
+<!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"><asteriskteam@digium.com></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>
\ No newline at end of file
--- /dev/null
+ 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(-)
projects / thirdparty / asterisk.git / commitdiff
