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297 lines
14 KiB
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<html><head><title>ChangeLog for asterisk-21.12.0</title></head><body>
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<h2>Change Log for Release asterisk-21.12.0</h2>
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<h3>Links:</h3>
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<ul>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html">Full ChangeLog</a> </li>
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<li><a href="https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0">GitHub Diff</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz">Tarball</a> </li>
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<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
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</ul>
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<h3>Summary:</h3>
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<ul>
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<li>Commits: 20</li>
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<li>Commit Authors: 10</li>
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<li>Issues Resolved: 13</li>
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<li>Security Advisories Resolved: 0</li>
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</ul>
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<h3>User Notes:</h3>
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<ul>
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<li>
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<h4>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</h4>
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<p>Added a new option to HANGUPCAUSE to access additional
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information about hangup reason. Reason headers from pjsip
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could be read using 'tech_extended' cause type.</p>
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</li>
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<li>
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<h4>chan_dahdi: Add DAHDI_CHANNEL function.</h4>
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<p>The DAHDI_CHANNEL function allows for getting/setting
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certain properties about DAHDI channels from the dialplan.</p>
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</li>
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</ul>
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<h3>Upgrade Notes:</h3>
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<ul>
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<li>
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<h4>res_audiosocket: add message types for all slin sample rates</h4>
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New audiosocket message types 0x11 - 0x18 has been added
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for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
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slin192 audio. External applications using audiosocket may need to be
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updated to support these message types if the audiosocket channel is
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created with one of these audio formats.</li>
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</ul>
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<h3>Developer Notes:</h3>
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<h3>Commit Authors:</h3>
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<ul>
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<li>Bastian Triller: (1)</li>
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<li>Ben Ford: (1)</li>
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<li>George Joseph: (4)</li>
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<li>Igor Goncharovsky: (1)</li>
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<li>Max Grobecker: (1)</li>
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<li>Nathan Monfils: (1)</li>
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<li>Naveen Albert: (4)</li>
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<li>Sean Bright: (3)</li>
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<li>Sven Kube: (3)</li>
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<li>phoneben: (1)</li>
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</ul>
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<h2>Issue and Commit Detail:</h2>
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<h3>Closed Issues:</h3>
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<ul>
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<li>1340: [bug]: comfort noise packet corrupted</li>
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<li>1419: [bug]: static code analysis issues in app_adsiprog.c</li>
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<li>1422: [bug]: static code analysis issues in apps/app_externalivr.c</li>
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<li>1425: [bug]: static code analysis issues in apps/app_queue.c</li>
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<li>1434: [improvement]: pbx_variables: Create real channel for dialplan eval CLI command</li>
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<li>1436: [improvement]: res_cliexec: Avoid unnecessary cast to char*</li>
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<li>1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function</li>
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<li>1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with PJSIP_TRANSFER_HANDLING(ari-only)</li>
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<li>1491: [bug]: Segfault: <code>channelstorage_cpp</code> fast lookup without lock (<code>get_by_name_exact</code>/<code>get_by_uniqueid</code>) leads to UAF during hangup</li>
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<li>1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message</li>
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<li>1539: [bug]: safe_asterisk without TTY doesn't log to file</li>
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<li>1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361</li>
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<li>1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend</li>
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</ul>
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<h3>Commits By Author:</h3>
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<ul>
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<li>
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<h4>Bastian Triller (1):</h4>
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</li>
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<li>
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<h4>Ben Ford (1):</h4>
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</li>
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<li>
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<h4>George Joseph (4):</h4>
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</li>
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<li>
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<h4>Igor Goncharovsky (1):</h4>
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</li>
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<li>
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<h4>Max Grobecker (1):</h4>
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</li>
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<li>
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<h4>Nathan Monfils (1):</h4>
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</li>
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<li>
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<h4>Naveen Albert (4):</h4>
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</li>
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<li>
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<h4>Sean Bright (3):</h4>
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</li>
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<li>
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<h4>Sven Kube (3):</h4>
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</li>
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<li>
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<h4>phoneben (1):</h4>
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</li>
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</ul>
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<h3>Commit List:</h3>
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<ul>
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<li>channelstorage: Allow storage driver read locking to be skipped.</li>
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<li>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</li>
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<li>safe_asterisk: Fix logging and sorting issue.</li>
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<li>res_audiosocket: add message types for all slin sample rates</li>
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<li>chan_websocket.c: Change payload references to command instead.</li>
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<li>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</li>
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<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</li>
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<li>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</li>
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<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
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<li>Fix some doxygen, typos and whitespace</li>
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<li>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create</li>
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<li>app_queue: Add NULL pointer checks in app_queue</li>
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<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
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<li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
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<li>app_adsiprog: Fix possible NULL dereference.</li>
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<li>manager.c: Fix presencestate object leak</li>
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<li>audiohook.c: Ensure correct AO2 reference is dereffed.</li>
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<li>res_cliexec: Remove unnecessary casts to char*.</li>
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<li>rtp_engine.c: Add exception for comfort noise payload.</li>
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<li>pbx_variables.c: Create real channel for "dialplan eval function".</li>
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</ul>
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<h3>Commit Details:</h3>
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<h4>channelstorage: Allow storage driver read locking to be skipped.</h4>
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<p>Author: George Joseph
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Date: 2025-11-06</p>
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<p>After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
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channels/externalMedia was called with a custom channel id AND the
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cpp_map_name_id channel storage backend is in use, a deadlock can occur when
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hanging up the channel. It's actually triggered in
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channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
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channelstorage driver then subsequently does a lookup for channel uniqueid
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which now does a read lock. This is an invalid operation and causes the lock
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state to get "bad". When the channels try to hang up, a write lock is
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attempted again which hangs and causes the deadlock.</p>
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<p>Now instead of the cpp_map_name_id channelstorage driver "get" APIs
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automatically performing a read lock, they take a "lock" parameter which
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allows a caller who already has a write lock to indicate that the "get" API
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must not attempt its own lock. This prevents the state from getting mesed up.</p>
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<p>The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
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have this issue but since it also implements the common channelstorage API,
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it needed its "get" implementations updated to take the lock parameter. They
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just don't use it.</p>
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<p>Resolves: #1578</p>
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<h4>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</h4>
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<p>Author: Sean Bright
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Date: 2025-10-22</p>
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<ul>
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<li>Using <code>==</code> with the POSIX sh <code>test</code> utility is UB.</li>
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<li>Switch back to using globs instead of using <code>$(find … | sort)</code>.</li>
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<li>Fix a missing redirect when checking for the OS type.</li>
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</ul>
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<p>Resolves: #1554</p>
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<h4>safe_asterisk: Fix logging and sorting issue.</h4>
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<p>Author: George Joseph
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Date: 2025-10-17</p>
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<p>Re-enabled "TTY=9" which was erroneously disabled as part of a recent
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security fix and removed another logging "fix" that was added.</p>
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<p>Also added a sort to the "find" that enumerates the scripts to be sourced so
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they're sourced in the correct order.</p>
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<p>Resolves: #1539</p>
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<h4>res_audiosocket: add message types for all slin sample rates</h4>
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<p>Author: Sven Kube
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Date: 2025-10-10</p>
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<p>Extend audiosocket messages with types 0x11 - 0x18 to create asterisk
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frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
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slin192 format, enabling the transmission of audio at a higher sample
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rates. For audiosocket messages sent by Asterisk, the message kind is
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determined by the format of the originating asterisk frame.</p>
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<p>UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added
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for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
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slin192 audio. External applications using audiosocket may need to be
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updated to support these message types if the audiosocket channel is
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created with one of these audio formats.</p>
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<h4>chan_websocket.c: Change payload references to command instead.</h4>
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<p>Author: George Joseph
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Date: 2025-10-08</p>
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<p>Some of the tests in process_text_message() were still comparing to the
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websocket message payload instead of the "command" string.</p>
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<p>Resolves: #1525</p>
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<h4>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</h4>
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<p>Author: Igor Goncharovsky
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Date: 2025-09-04</p>
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<p>As soon as SIP call may end with several Reason headers, we
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want to make all of them available through the HAGUPCAUSE() function.
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This implementation uses the same ao2 hash for cause codes storage
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and adds a flag to make difference between last processed sip
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message and content of reason headers.</p>
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<p>UserNote: Added a new option to HANGUPCAUSE to access additional
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information about hangup reason. Reason headers from pjsip
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could be read using 'tech_extended' cause type.</p>
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<h4>channelstorage_cpp_map_name_id: Add read locking around retrievals.</h4>
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<p>Author: George Joseph
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Date: 2025-10-01</p>
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<p>When we retrieve a channel from a C++ map, we actually get back a wrapper
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object that points to the channel then right after we retrieve it, we bump its
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reference count. There's a tiny chance however that between those two
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statements a delete and/or unref might happen which would cause the wrapper
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object or the channel itself to become invalid resulting in a SEGV. To avoid
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this we now perform a read lock on the driver around those statements.</p>
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<p>Resolves: #1491</p>
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<h4>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</h4>
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<p>Author: Max Grobecker
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Date: 2025-09-21</p>
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<p>This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation.
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The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header,
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but that option had no effect as it was not implemented by res_pjsip_geolocation.</p>
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<p>If the <code>location_source</code> configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).</p>
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<p>This commits adds already documented functionality.</p>
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<h4>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</h4>
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<p>Author: Sven Kube
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Date: 2025-09-22</p>
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<p>When handling SIP transfers via ARI, there is no protocol_id in case of
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a blind transfer.</p>
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<p>Resolves: #1467</p>
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<h4>Fix some doxygen, typos and whitespace</h4>
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<p>Author: Bastian Triller
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Date: 2025-09-21</p>
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<h4>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create</h4>
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<p>Author: Sven Kube
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Date: 2025-09-18</p>
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<p>When handling SIP transfers via ARI, the <code>referred_by</code> field in
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<code>transfer_ari_state</code> may be null, since SIP REFER requests are not
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required to include a <code>Referred-By</code> header. Without this check, a null
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value caused the transfer to fail and triggered a NOTIFY with a 500
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Internal Server Error.</p>
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<h4>app_queue: Add NULL pointer checks in app_queue</h4>
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<p>Author: phoneben
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Date: 2025-09-11</p>
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<p>Add NULL check for word_list before calling word_in_list()
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Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()</p>
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<p>Resolves: #1425</p>
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<h4>app_externalivr: Prevent out-of-bounds read during argument processing.</h4>
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<p>Author: Sean Bright
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Date: 2025-09-17</p>
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<p>Resolves: #1422</p>
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<h4>chan_dahdi: Add DAHDI_CHANNEL function.</h4>
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<p>Author: Naveen Albert
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Date: 2025-09-11</p>
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<p>Add a dialplan function that can be used to get/set properties of
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DAHDI channels (as opposed to Asterisk channels). This exposes
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properties that were not previously available, allowing for certain
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operations to now be performed in the dialplan.</p>
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<p>Resolves: #1455</p>
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<p>UserNote: The DAHDI_CHANNEL function allows for getting/setting
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certain properties about DAHDI channels from the dialplan.</p>
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<h4>app_adsiprog: Fix possible NULL dereference.</h4>
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<p>Author: Naveen Albert
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Date: 2025-09-10</p>
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<p>get_token can return NULL, but process_token uses this result without
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checking for NULL; as elsewhere, check for a NULL result to avoid
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possible NULL dereference.</p>
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<p>Resolves: #1419</p>
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<h4>manager.c: Fix presencestate object leak</h4>
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<p>Author: Nathan Monfils
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Date: 2025-09-08</p>
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<p>ast_presence_state allocates subtype and message. We straightforwardly
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need to clean those up.</p>
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<h4>audiohook.c: Ensure correct AO2 reference is dereffed.</h4>
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<p>Author: Sean Bright
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Date: 2025-09-10</p>
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<p>Part of #1440.</p>
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<h4>res_cliexec: Remove unnecessary casts to char*.</h4>
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<p>Author: Naveen Albert
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Date: 2025-09-09</p>
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<p>Resolves: #1436</p>
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<h4>rtp_engine.c: Add exception for comfort noise payload.</h4>
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<p>Author: Ben Ford
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Date: 2025-09-09</p>
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<p>In a previous commit, a change was made to
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ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample
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rates. This ended up returning an invalid payload int for comfort noise.
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A check has been added that returns early if the payload is in fact
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supposed to be comfort noise.</p>
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<p>Fixes: #1340</p>
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<h4>pbx_variables.c: Create real channel for "dialplan eval function".</h4>
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<p>Author: Naveen Albert
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Date: 2025-09-09</p>
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<p>"dialplan eval function" has been using a dummy channel for function
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evaluation, much like many of the unit tests. However, sometimes, this
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can cause issues for functions that are not expecting dummy channels.
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As an example, ast_channel_tech(chan) is NULL on such channels, and
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ast_channel_tech(chan)->type consequently results in a NULL dereference.
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Normally, functions do not worry about this since channels executing
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dialplan aren't dummy channels.</p>
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<p>While some functions are better about checking for these sorts of edge
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cases, use a real channel with a dummy technology to make this CLI
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command inherently safe for any dialplan function that could be evaluated
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from the CLI.</p>
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<p>Resolves: #1434</p>
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</body></html>
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