--- /dev/null
+<html><head><title>ChangeLog for asterisk-20.16.0-rc1</title></head><body>
+<h2>Change Log for Release asterisk-20.16.0-rc1</h2>
+<h3>Links:</h3>
+<ul>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0-rc1.html">Full ChangeLog</a> </li>
+<li><a href="https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0-rc1">GitHub Diff</a> </li>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0-rc1.tar.gz">Tarball</a> </li>
+<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
+</ul>
+<h3>Summary:</h3>
+<ul>
+<li>Commits: 51</li>
+<li>Commit Authors: 22</li>
+<li>Issues Resolved: 37</li>
+<li>Security Advisories Resolved: 0</li>
+</ul>
+<h3>User Notes:</h3>
+<ul>
+<li>
+<h4>app_queue.c: Add new global 'log_unpause_on_reason_change'</h4>
+<p>Add new global option 'log_unpause_on_reason_change' that
+ is default disabled. When enabled cause addition of UNPAUSE event on
+ every re-PAUSE with reason changed.</p>
+</li>
+<li>
+<h4>pbx_builtins: Allow custom tone for WaitExten.</h4>
+<p>The tone used while waiting for digits in WaitExten
+ can now be overridden by specifying an argument for the 'd'
+ option.</p>
+</li>
+<li>
+<h4>res_tonedetect: Add option for TONE_DETECT detection to auto stop.</h4>
+<p>The 'e' option for TONE_DETECT now allows detection to
+ be disabled automatically once the desired number of matches have
+ been fulfilled, which can help prevent race conditions in the
+ dialplan, since TONE_DETECT does not need to be disabled after
+ a hit.</p>
+</li>
+<li>
+<h4>sorcery: Prevent duplicate objects and ensure missing objects are created on u..</h4>
+<p>Users relying on Sorcery multiple writable backends configurations
+ (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
+ in sorcery.conf to ensure missing objects are recreated after temporary backend
+ failures. Default behavior remains unchanged unless explicitly enabled.</p>
+</li>
+<li>
+<h4>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</h4>
+<p>A new WebSocket channel driver option <code>v</code> has been added to the
+ Dial application that allows you to specify additional URI parameters on
+ outgoing connections. Run <code>core show application Dial</code> from the Asterisk CLI
+ to see how to use it.</p>
+</li>
+<li>
+<h4>app_chanspy: Add option to not automatically answer channel.</h4>
+<p>ChanSpy and ExtenSpy can now be configured to not
+ automatically answer the channel by using the 'N' option.</p>
+</li>
+<li>
+<h4>cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.</h4>
+<p>Enabling the tracking of the
+ STREAM_BEGIN and the STREAM_END event
+ types in cel.conf will log media files and
+ music on hold played to each channel.
+ The STREAM_BEGIN event's extra field will
+ contain a JSON with the file details (path,
+ format and language), or the class name, in
+ case of music on hold is played. The DTMF
+ event's extra field will contain a JSON with
+ the digit and the duration in milliseconds.</p>
+</li>
+<li>
+<h4>res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM</h4>
+<p>Options are now available in the menuselect "Resource Modules"
+ category that allow you to enable the AES_192, AES_256 and AES_GCM
+ cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
+ them but modern versions do. Previously, the only way to enable them was
+ to set the CFLAGS environment variable when running ./configure.
+ The default setting is to disable them preserving existing behavior.</p>
+</li>
+<li>
+<h4>cdr: add CANCEL dispostion in CDR</h4>
+<p>A new CDR option "canceldispositionenabled" has been added
+ that when set to true, the NO ANSWER disposition will be split into
+ two dispositions: CANCEL and NO ANSWER. The default value is 'no'</p>
+</li>
+<li>
+<h4>func_curl: Allow auth methods to be set.</h4>
+<p>The httpauth field in CURLOPT now allows the authentication
+ methods to be set.</p>
+</li>
+<li>
+<h4>Media over Websocket Channel Driver</h4>
+<p>A new channel driver "chan_websocket" is now available. It can
+ exchange media over both inbound and outbound websockets and will both frame
+ and re-time the media it receives.
+ See http://s.asterisk.net/mow for more information.
+ The ARI channels/externalMedia API now includes support for the</p>
+</li>
+</ul>
+<h3>Upgrade Notes:</h3>
+<h3>Developer Notes:</h3>
+<ul>
+<li>
+<h4>ARI: Add command to indicate progress to a channel</h4>
+<p>A new ARI endpoint is available at <code>/channels/{channelId}/progress</code> to indicate progress to a channel.</p>
+</li>
+<li>
+<h4>options: Change ast_options from ast_flags to ast_flags64.</h4>
+<p>The 32-bit ast_options has no room left to accomodate new
+ options and so has been converted to an ast_flags64 structure. All internal
+ references to ast_options have been updated to use the 64-bit flag
+ manipulation macros. External module references to the 32-bit ast_options
+ should continue to work on little-endian systems because the
+ least-significant bytes of a 64 bit integer will be in the same location as a
+ 32-bit integer. Because that's not the case on big-endian systems, we've
+ swapped the bytes in the flags manupulation macros on big-endian systems
+ so external modules should still work however you are encouraged to test.</p>
+</li>
+</ul>
+<h3>Commit Authors:</h3>
+<ul>
+<li>Alexei Gradinari: (2)</li>
+<li>Alexey Khabulyak: (2)</li>
+<li>Allan Nathanson: (1)</li>
+<li>Artem Umerov: (1)</li>
+<li>Ben Ford: (1)</li>
+<li>George Joseph: (9)</li>
+<li>Igor Goncharovsky: (2)</li>
+<li>Jaco Kroon: (1)</li>
+<li>Joe Garlick: (1)</li>
+<li>Jose Lopes: (1)</li>
+<li>Kodokaii: (1)</li>
+<li>Martin Tomec: (1)</li>
+<li>Mike Bradeen: (1)</li>
+<li>Mkmer: (1)</li>
+<li>Naveen Albert: (15)</li>
+<li>Sean Bright: (2)</li>
+<li>Sperl Viktor: (2)</li>
+<li>Stanislav Abramenkov: (1)</li>
+<li>Stuart Henderson: (1)</li>
+<li>Sven Kube: (2)</li>
+<li>Tinet-Mucw: (2)</li>
+<li>Zhou_jiajian: (1)</li>
+</ul>
+<h2>Issue and Commit Detail:</h2>
+<h3>Closed Issues:</h3>
+<ul>
+<li>401: [bug]: app_dial: Answer Gosub option passthrough regression</li>
+<li>927: [bug]: no audio when media source changed during the call</li>
+<li>1176: [bug]: ast_slinear_saturated_multiply_float produces potentially audible distortion artifacts</li>
+<li>1259: [bug]: New TenantID feature doesn't seem to set CDR for incoming calls</li>
+<li>1260: [bug]: Asterisk sends RTP audio stream before ICE/DTLS completes</li>
+<li>1269: [bug]: MixMonitor with D option produces corrupt file</li>
+<li>1273: [bug]: When executed with GotoIf, the action Redirect does not take effect and causes confusion in dialplan execution.</li>
+<li>1280: [improvement]: logging playback of audio per channel</li>
+<li>1289: [bug]: sorcery - duplicate objects from multiple backends and backend divergence on update</li>
+<li>1301: [bug]: sig_analog: fgccamamf doesn't handle STP, STP2, or STP3</li>
+<li>1304: [bug]: FLUSH_MEDIA does not reset frame_queue_length in WebSocket channel</li>
+<li>1305: [bug]: Realtime incorrectly falls back to next backend on record-not-found (SQL_NO_DATA), causing incorrect behavior and delay</li>
+<li>1307: [improvement]: ast_tls_cert: Allow certificate validity to be configurable</li>
+<li>1309: [bug]: Crash with C++ alternative storage backend enabled</li>
+<li>1315: [bug]: When executed with dialplan, the action Redirect does not take effect.</li>
+<li>1317: [bug]: AGI command buffer overflow with long variables</li>
+<li>1321: [improvement]: app_agent_pool: Remove obsolete documentation</li>
+<li>1323: [new-feature]: add CANCEL dispostion in CDR</li>
+<li>1327: [bug]: res_stasis_device_state: can't delete ARI Devicestate after asterisk restart</li>
+<li>1332: [new-feature]: func_curl: Allow auth methods to be set</li>
+<li>1349: [bug]: Race condition on redirect can cause missing Diversion header</li>
+<li>1352: [improvement]: Websocket channel with custom URI</li>
+<li>1353: [bug]: AST_DATA_DIR/sounds/custom directory not searched</li>
+<li>1358: [new-feature]: app_chanspy: Add option to not automatically answer channel</li>
+<li>1364: [bug]: bridge.c: BRIDGE_NOANSWER not always obeyed</li>
+<li>1366: [improvement]: func_frame_drop: Handle allocation failure properly</li>
+<li>1369: [bug]: test_res_prometheus: Compilation failure in devmode due to curlopts not using long type</li>
+<li>1371: [improvement]: func_frame_drop: Add debug messages for frames that can be dropped</li>
+<li>1375: [improvement]: dsp.c: Improve logging in tone_detect().</li>
+<li>1378: [bug]: chan_dahdi: dialmode feature is not properly reset between calls</li>
+<li>1380: [bug]: sig_analog: Segfault due to calling strcmp on NULL</li>
+<li>1384: [bug]: chan_websocket: asterisk crashes on hangup after STOP_MEDIA_BUFFERING command with id</li>
+<li>1386: [bug]: enabling announceposition_only_up prevents any queue position announcements</li>
+<li>1390: [improvement]: res_tonedetect: Add option to automatically end detection in TONE_DETECT</li>
+<li>1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled</li>
+<li>1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable</li>
+<li>1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage</li>
+</ul>
+<h3>Commits By Author:</h3>
+<ul>
+<li>
+<h4>Alexei Gradinari (2):</h4>
+</li>
+<li>res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)</li>
+<li>
+<p>sorcery: Prevent duplicate objects and ensure missing objects are created on u..</p>
+</li>
+<li>
+<h4>Alexey Khabulyak (2):</h4>
+</li>
+<li>app_dial.c: Moved channel lock to prevent deadlock</li>
+<li>
+<p>pbx_lua.c: segfault when pass null data to term_color function</p>
+</li>
+<li>
+<h4>Allan Nathanson (1):</h4>
+</li>
+<li>
+<p>file.c: with "sounds_search_custom_dir = yes", search "custom" directory</p>
+</li>
+<li>
+<h4>Artem Umerov (1):</h4>
+</li>
+<li>
+<p>Fix missing ast_test_flag64 in extconf.c</p>
+</li>
+<li>
+<h4>Ben Ford (1):</h4>
+</li>
+<li>
+<p>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</p>
+</li>
+<li>
+<h4>George Joseph (9):</h4>
+</li>
+<li>Media over Websocket Channel Driver</li>
+<li>app_mixmonitor: Update the documentation concerning the "D" option.</li>
+<li>cdr.c: Set tenantid from party_a->base instead of chan->base.</li>
+<li>options: Change ast_options from ast_flags to ast_flags64.</li>
+<li>res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM</li>
+<li>channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.</li>
+<li>xmldoc.c: Fix rendering of CLI output.</li>
+<li>chan_websocket: Fix buffer overrun when processing TEXT websocket frames.</li>
+<li>
+<p>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</p>
+</li>
+<li>
+<h4>Igor Goncharovsky (2):</h4>
+</li>
+<li>app_waitforsilence.c: Use milliseconds to calculate timeout time</li>
+<li>
+<p>app_queue.c: Add new global 'log_unpause_on_reason_change'</p>
+</li>
+<li>
+<h4>Jaco Kroon (1):</h4>
+</li>
+<li>
+<p>res_musiconhold: Appropriately lock channel during start.</p>
+</li>
+<li>
+<h4>Joe Garlick (1):</h4>
+</li>
+<li>
+<p>chan_websocket.c: Add DTMF messages</p>
+</li>
+<li>
+<h4>Jose Lopes (1):</h4>
+</li>
+<li>
+<p>res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.</p>
+</li>
+<li>
+<h4>Martin Tomec (1):</h4>
+</li>
+<li>
+<p>chan_pjsip.c: Change SSRC after media source change</p>
+</li>
+<li>
+<h4>Mike Bradeen (1):</h4>
+</li>
+<li>
+<p>res_pjsip_diversion: resolve race condition between Diversion header processin..</p>
+</li>
+<li>
+<h4>Naveen Albert (15):</h4>
+</li>
+<li>sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.</li>
+<li>ast_tls_cert: Make certificate validity configurable.</li>
+<li>app_agent_pool: Remove documentation for removed option.</li>
+<li>func_curl: Allow auth methods to be set.</li>
+<li>bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.</li>
+<li>func_frame_drop: Handle allocation failure properly.</li>
+<li>test_res_prometheus: Fix compilation failure on Debian 13.</li>
+<li>func_frame_drop: Add debug messages for dropped frames.</li>
+<li>app_chanspy: Add option to not automatically answer channel.</li>
+<li>dsp.c: Improve debug logging in tone_detect().</li>
+<li>sig_analog: Fix SEGV due to calling strcmp on NULL.</li>
+<li>chan_dahdi: Fix erroneously persistent dialmode.</li>
+<li>sig_analog: Skip Caller ID spill if usecallerid=no.</li>
+<li>res_tonedetect: Add option for TONE_DETECT detection to auto stop.</li>
+<li>
+<p>pbx_builtins: Allow custom tone for WaitExten.</p>
+</li>
+<li>
+<h4>Sean Bright (2):</h4>
+</li>
+<li>res_musiconhold.c: Annotate when the channel is locked.</li>
+<li>
+<p>res_musiconhold.c: Ensure we're always locked around music state access.</p>
+</li>
+<li>
+<h4>Sperl Viktor (2):</h4>
+</li>
+<li>res_agi: Increase AGI command buffer size from 2K to 8K</li>
+<li>
+<p>cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.</p>
+</li>
+<li>
+<h4>Stanislav Abramenkov (1):</h4>
+</li>
+<li>
+<p>bundled_pjproject: Avoid deadlock between transport and transaction</p>
+</li>
+<li>
+<h4>Stuart Henderson (1):</h4>
+</li>
+<li>
+<p>app_queue: fix comparison for announce-position-only-up</p>
+</li>
+<li>
+<h4>Sven Kube (2):</h4>
+</li>
+<li>resource_channels.c: Don't call ast_channel_get_by_name on empty optional argu..</li>
+<li>
+<p>ARI: Add command to indicate progress to a channel</p>
+</li>
+<li>
+<h4>Tinet-mucw (2):</h4>
+</li>
+<li>pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return..</li>
+<li>
+<p>pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper sho..</p>
+</li>
+<li>
+<h4>kodokaii (1):</h4>
+</li>
+<li>
+<p>chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA</p>
+</li>
+<li>
+<h4>mkmer (1):</h4>
+</li>
+<li>
+<p>utils.h: Add rounding to float conversion to int.</p>
+</li>
+<li>
+<h4>zhou_jiajian (1):</h4>
+</li>
+<li>cdr: add CANCEL dispostion in CDR</li>
+</ul>
+<h3>Commit List:</h3>
+<ul>
+<li>chan_websocket.c: Add DTMF messages</li>
+<li>app_queue.c: Add new global 'log_unpause_on_reason_change'</li>
+<li>app_waitforsilence.c: Use milliseconds to calculate timeout time</li>
+<li>Fix missing ast_test_flag64 in extconf.c</li>
+<li>pbx_builtins: Allow custom tone for WaitExten.</li>
+<li>res_tonedetect: Add option for TONE_DETECT detection to auto stop.</li>
+<li>app_queue: fix comparison for announce-position-only-up</li>
+<li>sig_analog: Skip Caller ID spill if usecallerid=no.</li>
+<li>chan_dahdi: Fix erroneously persistent dialmode.</li>
+<li>chan_websocket: Fix buffer overrun when processing TEXT websocket frames.</li>
+<li>sig_analog: Fix SEGV due to calling strcmp on NULL.</li>
+<li>ARI: Add command to indicate progress to a channel</li>
+<li>dsp.c: Improve debug logging in tone_detect().</li>
+<li>res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.</li>
+<li>app_chanspy: Add option to not automatically answer channel.</li>
+<li>xmldoc.c: Fix rendering of CLI output.</li>
+<li>func_frame_drop: Add debug messages for dropped frames.</li>
+<li>test_res_prometheus: Fix compilation failure on Debian 13.</li>
+<li>func_frame_drop: Handle allocation failure properly.</li>
+<li>pbx_lua.c: segfault when pass null data to term_color function</li>
+<li>bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.</li>
+<li>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</li>
+<li>app_dial.c: Moved channel lock to prevent deadlock</li>
+<li>file.c: with "sounds_search_custom_dir = yes", search "custom" directory</li>
+<li>cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.</li>
+<li>channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.</li>
+<li>res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM</li>
+<li>cdr: add CANCEL dispostion in CDR</li>
+<li>func_curl: Allow auth methods to be set.</li>
+<li>options: Change ast_options from ast_flags to ast_flags64.</li>
+<li>res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)</li>
+<li>app_agent_pool: Remove documentation for removed option.</li>
+<li>res_agi: Increase AGI command buffer size from 2K to 8K</li>
+<li>ast_tls_cert: Make certificate validity configurable.</li>
+<li>cdr.c: Set tenantid from party_a->base instead of chan->base.</li>
+<li>app_mixmonitor: Update the documentation concerning the "D" option.</li>
+<li>sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.</li>
+<li>chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA</li>
+<li>chan_pjsip.c: Change SSRC after media source change</li>
+<li>Media over Websocket Channel Driver</li>
+<li>bundled_pjproject: Avoid deadlock between transport and transaction</li>
+<li>utils.h: Add rounding to float conversion to int.</li>
+<li>res_musiconhold.c: Ensure we're always locked around music state access.</li>
+<li>res_musiconhold.c: Annotate when the channel is locked.</li>
+<li>res_musiconhold: Appropriately lock channel during start.</li>
+</ul>
+<h3>Commit Details:</h3>
+<h4>chan_websocket.c: Add DTMF messages</h4>
+<p>Author: Joe Garlick
+ Date: 2025-09-04</p>
+<p>Added DTMF messages to the chan_websocket feature.</p>
+<p>When a user presses DTMF during a call over chan_websocket it will send a message like:
+ "DTMF_END digit:1"</p>
+<p>Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70</p>
+<h4>app_queue.c: Add new global 'log_unpause_on_reason_change'</h4>
+<p>Author: Igor Goncharovsky
+ Date: 2025-09-02</p>
+<p>In many asterisk-based systems, the pause reason is used to separate
+ pauses by type,and logically, changing the reason defines two intervals
+ that should be accounted for separately. The introduction of a new
+ option allows me to separate the intervals of operator inactivity in
+ the log by the event of unpausing.</p>
+<p>UserNote: Add new global option 'log_unpause_on_reason_change' that
+ is default disabled. When enabled cause addition of UNPAUSE event on
+ every re-PAUSE with reason changed.</p>
+<h4>app_waitforsilence.c: Use milliseconds to calculate timeout time</h4>
+<p>Author: Igor Goncharovsky
+ Date: 2025-09-04</p>
+<p>The functions WaitForNoise() and WaitForSilence() use the time()
+ functions to calculate elapsed time, which causes the timer to fire on
+ a whole second boundary, and the actual function execution time to fire
+ the timer may be 1 second less than expected. This fix replaces time()
+ with ast_tvnow().</p>
+<p>Fixes: #1401</p>
+<h4>Fix missing ast_test_flag64 in extconf.c</h4>
+<p>Author: Artem Umerov
+ Date: 2025-08-29</p>
+<p>Fix missing ast_test_flag64 after https://github.com/asterisk/asterisk/commit/43bf8a4ded7a65203b766b91eaf8331a600e9d8d</p>
+<h4>pbx_builtins: Allow custom tone for WaitExten.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-25</p>
+<p>Currently, the 'd' option will play dial tone while waiting
+ for digits. Allow it to accept an argument for any tone from
+ indications.conf.</p>
+<p>Resolves: #1396</p>
+<p>UserNote: The tone used while waiting for digits in WaitExten
+ can now be overridden by specifying an argument for the 'd'
+ option.</p>
+<h4>res_tonedetect: Add option for TONE_DETECT detection to auto stop.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-28</p>
+<p>One of the problems with TONE_DETECT as it was originally written
+ is that if a tone is detected multiple times, it can trigger
+ the redirect logic multiple times as well. For example, if we
+ do an async goto in the dialplan after detecting a tone, because
+ the detector is still active until explicitly disabled, if we
+ detect the tone again, we will branch again and start executing
+ that dialplan a second time. This is rarely ever desired behavior,
+ and can happen if the detector is not removed quickly enough.</p>
+<p>Add a new option, 'e', which automatically disables the detector
+ once the desired number of matches have been heard. This eliminates
+ the potential race condition where previously the detector would
+ need to be disabled immediately, but doing so quickly enough
+ was not guaranteed. This also allows match criteria to be retained
+ longer if needed, so the detector does not need to be destroyed
+ prematurely.</p>
+<p>Resolves: #1390</p>
+<p>UserNote: The 'e' option for TONE_DETECT now allows detection to
+ be disabled automatically once the desired number of matches have
+ been fulfilled, which can help prevent race conditions in the
+ dialplan, since TONE_DETECT does not need to be disabled after
+ a hit.</p>
+<h4>app_queue: fix comparison for announce-position-only-up</h4>
+<p>Author: Stuart Henderson
+ Date: 2025-08-21</p>
+<p>Numerically comparing that the current queue position is less than
+ last_pos_said can only be done after at least one announcement has been
+ made, otherwise last_pos_said is at the default (0).</p>
+<p>Fixes: #1386</p>
+<h4>sorcery: Prevent duplicate objects and ensure missing objects are created on u..</h4>
+<p>Author: Alexei Gradinari
+ Date: 2025-07-07</p>
+<p>This patch resolves two issues in Sorcery objectset handling with multiple
+ backends:</p>
+<ol>
+<li>
+<p>Prevent duplicate objects:
+ When an object exists in more than one backend (e.g., a contact in both
+ 'astdb' and 'realtime'), the objectset previously returned multiple instances
+ of the same logical object. This caused logic failures in components like the
+ PJSIP registrar, where duplicate contact entries led to overcounting and
+ incorrect deletions, when max_contacts=1 and remove_existing=yes.</p>
+<p>This patch ensures only one instance of an object with a given key is added
+ to the objectset, avoiding these duplicate-related side effects.</p>
+</li>
+<li>
+<p>Ensure missing objects are created:
+ When using multiple writable backends, a temporary backend failure can lead
+ to objects missing permanently from that backend.
+ Currently, .update() silently fails if the object is not present,
+ and no .create() is attempted.
+ This results in inconsistent state across backends (e.g. astdb vs. realtime).</p>
+<p>This patch introduces a new global option in sorcery.conf:
+ [general]
+ update_or_create_on_update_miss = yes|no</p>
+<p>Default: no (preserves existing behavior).</p>
+<p>When enabled: if .update() fails with no data found, .create() is attempted
+ in that backend. This ensures that objects missing due to temporary backend
+ outages are re-synchronized once the backend is available again.</p>
+<p>Added a new CLI command:
+ sorcery show settings
+ Displays global Sorcery settings, including the current value of
+ update_or_create_on_update_miss.</p>
+<p>Updated tests to validate both flag enabled/disabled behavior.</p>
+</li>
+</ol>
+<p>Fixes: #1289</p>
+<p>UserNote: Users relying on Sorcery multiple writable backends configurations
+ (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
+ in sorcery.conf to ensure missing objects are recreated after temporary backend
+ failures. Default behavior remains unchanged unless explicitly enabled.</p>
+<h4>sig_analog: Skip Caller ID spill if usecallerid=no.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-25</p>
+<p>If Caller ID is disabled for an FXS port, then we should not send any
+ Caller ID spill on the line, as we have no Caller ID information that
+ we can/should be sending.</p>
+<p>Resolves: #1394</p>
+<h4>chan_dahdi: Fix erroneously persistent dialmode.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-18</p>
+<p>It is possible to modify the dialmode setting in the chan_dahdi/sig_analog
+ private using the CHANNEL function, to modify it during calls. However,
+ it was not being reset between calls, meaning that if, for example, tone
+ dialing was disabled, it would never work again unless explicitly enabled.</p>
+<p>This fixes the setting by pairing it with a "perm" version of the setting,
+ as a few other features have, so that it can be reset to the permanent
+ setting between calls. The documentation is also clarified to explain
+ the interaction of this setting and the digitdetect setting more clearly.</p>
+<p>Resolves: #1378</p>
+<h4>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</h4>
+<p>Author: George Joseph
+ Date: 2025-08-13</p>
+<ul>
+<li>Added a new option to the WebSocket dial string to capture the additional
+ URI parameters.</li>
+<li>Added a new API ast_uri_verify_encoded() that verifies that a string
+ either doesn't need URI encoding or that it has already been encoded.</li>
+<li>Added a new API ast_websocket_client_add_uri_params() to add the params
+ to the client websocket session.</li>
+<li>Added XML documentation that will show up with <code>core show application Dial</code>
+ that shows how to use it.</li>
+</ul>
+<p>Resolves: #1352</p>
+<p>UserNote: A new WebSocket channel driver option <code>v</code> has been added to the
+ Dial application that allows you to specify additional URI parameters on
+ outgoing connections. Run <code>core show application Dial</code> from the Asterisk CLI
+ to see how to use it.</p>
+<h4>chan_websocket: Fix buffer overrun when processing TEXT websocket frames.</h4>
+<p>Author: George Joseph
+ Date: 2025-08-19</p>
+<p>ast_websocket_read() receives data into a fixed 64K buffer then continually
+ reallocates a final buffer that, after all continuation frames have been
+ received, is the exact length of the data received and returns that to the
+ caller. process_text_message() in chan_websocket was attempting to set a
+ NULL terminator on the received payload assuming the payload buffer it
+ received was the large 64K buffer. The assumption was incorrect so when it
+ tried to set a NULL terminator on the payload, it could, depending on the
+ state of the heap at the time, cause heap corruption.</p>
+<p>process_text_message() now allocates its own payload_len + 1 sized buffer,
+ copies the payload received from ast_websocket_read() into it then NULL
+ terminates it prevent the possibility of the overrun and corruption.</p>
+<p>Resolves: #1384</p>
+<h4>sig_analog: Fix SEGV due to calling strcmp on NULL.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-18</p>
+<p>Add an additional check to guard against the channel application being
+ NULL.</p>
+<p>Resolves: #1380</p>
+<h4>ARI: Add command to indicate progress to a channel</h4>
+<p>Author: Sven Kube
+ Date: 2025-07-30</p>
+<p>Adds an ARI command to send a progress indication to a channel.</p>
+<p>DeveloperNote: A new ARI endpoint is available at <code>/channels/{channelId}/progress</code> to indicate progress to a channel.</p>
+<h4>dsp.c: Improve debug logging in tone_detect().</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-15</p>
+<p>The debug logging during DSP processing has always been kind
+ of overwhelming and annoying to troubleshoot. Simplify and
+ improve the logging in a few ways to aid DSP debugging:</p>
+<ul>
+<li>If we had a DSP hit, don't also emit the previous debug message that
+ was always logged. It is duplicated by the hit message, so this can
+ reduce the number of debug messages during detection by 50%.</li>
+<li>Include the hit count and required number of hits in the message so
+ on partial detections can be more easily troubleshot.</li>
+<li>Use debug level 9 for hits instead of 10, so we can focus on hits
+ without all the noise from the per-frame debug message.</li>
+<li>1-index the hit count in the debug messages. On the first hit, it
+ currently logs '0', just as when we are not detecting anything,
+ which can be confusing.</li>
+</ul>
+<p>Resolves: #1375</p>
+<h4>res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.</h4>
+<p>Author: Jose Lopes
+ Date: 2025-07-30</p>
+<p>After an asterisk restart, the deletion of ARI Devicestates didn't
+ return error, but the devicestate was not deleted.
+ Found a typo on populate_cache function that created wrong cache for
+ device states.
+ This bug caused wrong assumption that devicestate didn't exist,
+ since it was not in cache, so deletion didn't returned error.</p>
+<p>Fixes: #1327</p>
+<h4>app_chanspy: Add option to not automatically answer channel.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-13</p>
+<p>Add an option for ChanSpy and ExtenSpy to not answer the channel
+ automatically. Most applications that auto-answer by default
+ already have an option to disable this behavior if unwanted.</p>
+<p>Resolves: #1358</p>
+<p>UserNote: ChanSpy and ExtenSpy can now be configured to not
+ automatically answer the channel by using the 'N' option.</p>
+<h4>xmldoc.c: Fix rendering of CLI output.</h4>
+<p>Author: George Joseph
+ Date: 2025-08-14</p>
+<p>If you do a <code>core show application Dial</code>, you'll see it's kind of a mess.
+ Indents are wrong is some places, examples are printed in black which makes
+ them invisible on most terminals, and the lack of line breaks in some cases
+ makes it hard to follow.</p>
+<ul>
+<li>Fixed the rendering of examples so they are indented properly and changed
+ the color so they can be seen.</li>
+<li>There is now a line break before each option.</li>
+<li>Options are now printed on their own line with all option content indented
+ below them.</li>
+</ul>
+<p>Example from Dial before fixes:
+ ```
+ Example: Dial 555-1212 on first available channel in group 1, searching
+ from highest to lowest</p>
+<pre><code> Example: Ringing FXS channel 4 with ring cadence 2
+
+ Example: Dial 555-1212 on channel 3 and require answer confirmation
+</code></pre>
+<p>...</p>
+<pre><code> O([mode]):
+ mode - With <mode> either not specified or set to '1', the originator
+ hanging up will cause the phone to ring back immediately.
+</code></pre>
+<ul>
+<li>
+<p>With <mode> set to '2', when the operator flashes the trunk, it will ring
+ their phone back.
+ Enables <em>operator services</em> mode. This option only works when bridging a DAHDI
+ channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it
+ will be ignored. When the destination answers (presumably an operator services
+ station), the originator no longer has control of their line. They may hang up,
+ but the switch will not release their line until the destination party (the
+ operator) hangs up.</p>
+<p>p: This option enables screening mode. This is basically Privacy mode
+ without memory.
+ ```</p>
+</li>
+</ul>
+<p>After:
+ ```
+ Example: Dial 555-1212 on first available channel in group 1, searching
+ from highest to lowest</p>
+<pre><code> same => n,Dial(DAHDI/g1/5551212)
+
+ Example: Ringing FXS channel 4 with ring cadence 2
+
+ same => n,Dial(DAHDI/4r2)
+
+ Example: Dial 555-1212 on channel 3 and require answer confirmation
+
+ same => n,Dial(DAHDI/3c/5551212)
+</code></pre>
+<p>...</p>
+<pre><code> O([mode]):
+ mode - With <mode> either not specified or set to '1', the originator
+ hanging up will cause the phone to ring back immediately.
+ With <mode> set to '2', when the operator flashes the trunk, it will
+ ring their phone back.
+ Enables *operator services* mode. This option only works when bridging
+ a DAHDI channel to another DAHDI channel only. If specified on
+ non-DAHDI interfaces, it will be ignored. When the destination answers
+ (presumably an operator services station), the originator no longer has
+ control of their line. They may hang up, but the switch will not
+ release their line until the destination party (the operator) hangs up.
+
+ p:
+ This option enables screening mode. This is basically Privacy mode
+ without memory.
+</code></pre>
+<p>```</p>
+<p>There are still things we can do to make this more readable but this is a
+ start.</p>
+<h4>func_frame_drop: Add debug messages for dropped frames.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-14</p>
+<p>Add debug messages in scenarios where frames that are usually processed
+ are dropped or skipped.</p>
+<p>Resolves: #1371</p>
+<h4>test_res_prometheus: Fix compilation failure on Debian 13.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-14</p>
+<p>curl_easy_setopt expects long types, so be explicit.</p>
+<p>Resolves: #1369</p>
+<h4>func_frame_drop: Handle allocation failure properly.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-14</p>
+<p>Handle allocation failure and simplify the allocation using asprintf.</p>
+<p>Resolves: #1366</p>
+<h4>pbx_lua.c: segfault when pass null data to term_color function</h4>
+<p>Author: Alexey Khabulyak
+ Date: 2025-08-14</p>
+<p>This can be reproduced under certain curcomstences.
+ For example: call app.playback from lua with invalid data: app.playback({}).
+ pbx_lua.c will try to get data for this playback using lua_tostring function.
+ This function returs NULL for everything but strings and numbers.
+ Then, it calls term_color with NULL data.
+ term_color function can call(if we don't use vt100 compat term)
+ ast_copy_string with NULL inbuf which cause segfault. bt example:
+ ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0)
+ at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412</p>
+<p>Resolves: https://github.com/asterisk/asterisk/issues/1363</p>
+<h4>bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-14</p>
+<p>If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed
+ to answer when another channel bridges to it using Bridge(), and this is
+ checked when ast_bridge_call* is called. However, another path exists
+ (bridge_exec -> ast_bridge_add_channel) where this variable was not
+ checked and channels would be answered. We now check the variable there.</p>
+<p>Resolves: #401
+ Resolves: #1364</p>
+<h4>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</h4>
+<p>Author: Ben Ford
+ Date: 2025-08-04</p>
+<p>There was no check in __rtp_sendto that prevented Asterisk from sending
+ RTP before DTLS had finished negotiating. This patch adds logic to do
+ so.</p>
+<p>Fixes: #1260</p>
+<h4>app_dial.c: Moved channel lock to prevent deadlock</h4>
+<p>Author: Alexey Khabulyak
+ Date: 2025-08-04</p>
+<p>It's reproducible with pbx_lua, not regular dialplan.</p>
+<p>deadlock description:
+ 1. asterisk locks a channel
+ 2. calls function onedigit_goto
+ 3. calls ast_goto_if_exists funciton
+ 4. checks ast_exists_extension -> pbx_extension_helper
+ 5. pbx_extension_helper calls pbx_find_extension
+ 6. Then asterisk starts autoservice in a new thread
+ 7. autoservice run tries to lock the channel again</p>
+<p>Because our channel is locked already, autoservice can't lock.
+ Autoservice can't lock -> autoservice stop is waiting forever.
+ onedigit_goto waits for autoservice stop.</p>
+<p>Resolves: https://github.com/asterisk/asterisk/issues/1335</p>
+<h4>res_pjsip_diversion: resolve race condition between Diversion header processin..</h4>
+<p>Author: Mike Bradeen
+ Date: 2025-08-07</p>
+<p>Based on the firing order of the PJSIP call-backs on a redirect, it was possible for
+ the Diversion header to not be included in the outgoing 181 response to the UAC and
+ the INVITE to the UAS.</p>
+<p>This change moves the Diversion header processing to an earlier PJSIP callback while also
+ preventing the corresponding update that can cause a duplicate 181 response when processing
+ the header at that time.</p>
+<p>Resolves: #1349</p>
+<h4>file.c: with "sounds_search_custom_dir = yes", search "custom" directory</h4>
+<p>Author: Allan Nathanson
+ Date: 2025-08-10</p>
+<p>With <code>sounds_search_custom_dir = yes</code>, we are supposed to search for sounds
+ in the <code>AST_DATA_DIR/sounds/custom</code> directory before searching the normal
+ directories. Unfortunately, a recent change
+ (https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in
+ the "custom" directory not being searched. This change restores this
+ expected behavior.</p>
+<p>Resolves: #1353</p>
+<h4>cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.</h4>
+<p>Author: Sperl Viktor
+ Date: 2025-06-30</p>
+<p>Fixes: #1280</p>
+<p>UserNote: Enabling the tracking of the
+ STREAM_BEGIN and the STREAM_END event
+ types in cel.conf will log media files and
+ music on hold played to each channel.
+ The STREAM_BEGIN event's extra field will
+ contain a JSON with the file details (path,
+ format and language), or the class name, in
+ case of music on hold is played. The DTMF
+ event's extra field will contain a JSON with
+ the digit and the duration in milliseconds.</p>
+<h4>channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.</h4>
+<p>Author: George Joseph
+ Date: 2025-07-30</p>
+<p>The fact that deleting an object from a map invalidates any iterator
+ that happens to currently point to that object was overlooked in the initial
+ implementation. Unfortunately, there's no way to detect that an iterator
+ has been invalidated so the result was an occasional SEGV triggered by modules
+ like app_chanspy that opens an iterator and can keep it open for a long period
+ of time. The new implementation doesn't keep the underlying C++ iterator
+ open across calls to ast_channel_iterator_next() and uses a read lock
+ on the map to ensure that, even for the few microseconds we use the
+ iterator, another thread can't delete a channel from under it. Even with
+ this change, the iterators are still WAY faster than the ao2_legacy
+ storage driver.</p>
+<p>Full details about the new implementation are located in the comments for
+ iterator_next() in channelstorage_cpp_map_name_id.cc.</p>
+<p>Resolves: #1309</p>
+<h4>res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM</h4>
+<p>Author: George Joseph
+ Date: 2025-08-05</p>
+<p>UserNote: Options are now available in the menuselect "Resource Modules"
+ category that allow you to enable the AES_192, AES_256 and AES_GCM
+ cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
+ them but modern versions do. Previously, the only way to enable them was
+ to set the CFLAGS environment variable when running ./configure.
+ The default setting is to disable them preserving existing behavior.</p>
+<h4>cdr: add CANCEL dispostion in CDR</h4>
+<p>Author: zhou_jiajian
+ Date: 2025-07-24</p>
+<p>In the original implementation, both CANCEL and NO ANSWER states were
+ consolidated under the NO ANSWER disposition. This patch introduces a
+ separate CANCEL disposition, with an optional configuration switch to
+ enable this new disposition.</p>
+<p>Resolves: #1323</p>
+<p>UserNote: A new CDR option "canceldispositionenabled" has been added
+ that when set to true, the NO ANSWER disposition will be split into
+ two dispositions: CANCEL and NO ANSWER. The default value is 'no'</p>
+<h4>func_curl: Allow auth methods to be set.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-08-01</p>
+<p>Currently the CURL function only supports Basic Authentication,
+ the default auth method in libcurl. Add an option that also
+ allows enabling digest authentication.</p>
+<p>Resolves: #1332</p>
+<p>UserNote: The httpauth field in CURLOPT now allows the authentication
+ methods to be set.</p>
+<h4>options: Change ast_options from ast_flags to ast_flags64.</h4>
+<p>Author: George Joseph
+ Date: 2025-07-21</p>
+<p>DeveloperNote: The 32-bit ast_options has no room left to accomodate new
+ options and so has been converted to an ast_flags64 structure. All internal
+ references to ast_options have been updated to use the 64-bit flag
+ manipulation macros. External module references to the 32-bit ast_options
+ should continue to work on little-endian systems because the
+ least-significant bytes of a 64 bit integer will be in the same location as a
+ 32-bit integer. Because that's not the case on big-endian systems, we've
+ swapped the bytes in the flags manupulation macros on big-endian systems
+ so external modules should still work however you are encouraged to test.</p>
+<h4>res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)</h4>
+<p>Author: Alexei Gradinari
+ Date: 2025-07-15</p>
+<p>This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly
+ falls back to the next configured backend when the current one returns
+ SQL_NO_DATA (i.e., no record found).
+ This is a logical error and performance risk in multi-backend configurations.</p>
+<p>Solution:
+ Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker.
+ ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found.
+ Core engine stops iterating on this marker, avoiding unnecessary fallback.</p>
+<p>Notes:
+ Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly.
+ This patch only covers ODBC.</p>
+<p>Fixes: #1305</p>
+<h4>resource_channels.c: Don't call ast_channel_get_by_name on empty optional argu..</h4>
+<p>Author: Sven Kube
+ Date: 2025-07-30</p>
+<p><code>ast_ari_channels_create</code> and <code>ast_ari_channels_dial</code> called the
+ <code>ast_channel_get_by_name</code> function with optional arguments. Since
+ 8f1982c4d6, this function logs an error for empty channel names.
+ This commit adds checks for empty optional arguments that are used
+ to call <code>ast_channel_get_by_name</code> to prevent these error logs.</p>
+<h4>app_agent_pool: Remove documentation for removed option.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-07-28</p>
+<p>The already-deprecated "password" option for the AGENT function was
+ removed in commit d43b17a872e8227aa8a9905a21f90bd48f9d5348 for
+ Asterisk 12, but the documentation for it wasn't removed then.</p>
+<p>Resolves: #1321</p>
+<h4>pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper sho..</h4>
+<p>Author: Tinet-mucw
+ Date: 2025-07-22</p>
+<p>Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect.
+ In the situation that action Redirect is broken by pbx_extension_helper this info is changed.
+ This will cause the current dialplan location to be executed twice.
+ In other words, the Redirect action does not take effect.</p>
+<p>Resolves: #1315</p>
+<h4>res_agi: Increase AGI command buffer size from 2K to 8K</h4>
+<p>Author: Sperl Viktor
+ Date: 2025-07-22</p>
+<p>Fixes: #1317</p>
+<h4>ast_tls_cert: Make certificate validity configurable.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-07-16</p>
+<p>Currently, the ast_tls_cert script is hardcoded to produce certificates
+ with a validity of 365 days, which is not generally desirable for self-
+ signed certificates. Make this parameter configurable.</p>
+<p>Resolves: #1307</p>
+<h4>cdr.c: Set tenantid from party_a->base instead of chan->base.</h4>
+<p>Author: George Joseph
+ Date: 2025-07-17</p>
+<p>The CDR tenantid was being set in cdr_object_alloc from the channel->base
+ snapshot. Since this happens at channel creation before the dialplan is even
+ reached, calls to <code>CHANNEL(tenantid)=<something></code> in the dialplan were being
+ ignored. Instead we now take tenantid from party_a when
+ cdr_object_create_public_records() is called which is after the call has
+ ended and all channel snapshots rebuilt. This is exactly how accountcode
+ and amaflags, which can also be set in tha dialplpan, are handled.</p>
+<p>Resolves: #1259</p>
+<h4>app_mixmonitor: Update the documentation concerning the "D" option.</h4>
+<p>Author: George Joseph
+ Date: 2025-07-16</p>
+<p>When using the "D" option to output interleaved audio, the file extension
+ must be ".raw". That info wasn't being properly rendered in the markdown
+ and HTML on the documentation site. The XML was updated to move the
+ note in the option section to a warning in the description.</p>
+<p>Resolves: #1269</p>
+<h4>sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.</h4>
+<p>Author: Naveen Albert
+ Date: 2025-07-14</p>
+<p>Previously, we were only using # (ST) as a terminator, and not handling
+ A (STP), B (ST2P), or C (ST3P), which erroneously led to it being
+ treated as part of the dialed number. Parse any of these as the start
+ digit.</p>
+<p>Resolves: #1301</p>
+<h4>chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA</h4>
+<p>Author: kodokaii
+ Date: 2025-07-03</p>
+<p>In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from
+ the queue but does not reset the frame_queue_length counter.</p>
+<p>As a result, the driver incorrectly thinks the queue is full after flushing,
+ which prevents new multimedia frames from being sent, especially after multiple
+ flush commands.</p>
+<p>This fix sets frame_queue_length to 0 after flushing, ensuring the queue state
+ is consistent with its actual content.</p>
+<p>Fixes: #1304</p>
+<h4>chan_pjsip.c: Change SSRC after media source change</h4>
+<p>Author: Martin Tomec
+ Date: 2025-06-25</p>
+<p>When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly.
+ To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream.
+ This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.</p>
+<p>Resolves: #927</p>
+<h4>Media over Websocket Channel Driver</h4>
+<p>Author: George Joseph
+ Date: 2025-04-28</p>
+<ul>
+<li>
+<p>Created chan_websocket which can exchange media over both inbound and
+ outbound websockets which the driver will frame and time.
+ See http://s.asterisk.net/mow for more information.</p>
+</li>
+<li>
+<p>res_http_websocket: Made defines for max message size public and converted
+ a few nuisance verbose messages to debugs.</p>
+</li>
+<li>
+<p>main/channel.c: Changed an obsolete nuisance error to a debug.</p>
+</li>
+<li>
+<p>ARI channels: Updated externalMedia to include chan_websocket as a supported
+ transport.</p>
+</li>
+</ul>
+<p>UserNote: A new channel driver "chan_websocket" is now available. It can
+ exchange media over both inbound and outbound websockets and will both frame
+ and re-time the media it receives.
+ See http://s.asterisk.net/mow for more information.</p>
+<p>UserNote: The ARI channels/externalMedia API now includes support for the
+ WebSocket transport provided by chan_websocket.</p>
+<h4>bundled_pjproject: Avoid deadlock between transport and transaction</h4>
+<p>Author: Stanislav Abramenkov
+ Date: 2025-07-01</p>
+<p>Backport patch from upstream
+ * Avoid deadlock between transport and transaction
+ https://github.com/pjsip/pjproject/commit/edde06f261ac</p>
+<p>Issue described in
+ https://github.com/pjsip/pjproject/issues/4442</p>
+<h4>utils.h: Add rounding to float conversion to int.</h4>
+<p>Author: mkmer
+ Date: 2025-03-23</p>
+<p>Quote from an audio engineer NR9V:
+ There is a minor issue of a small amount of crossover distortion though as a result of <code>ast_slinear_saturated_multiply_float()</code> not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion.</p>
+<p>Resolves: #1176</p>
+<h4>pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return..</h4>
+<p>Author: Tinet-mucw
+ Date: 2025-06-18</p>
+<p>Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect.
+ In the situation that action Redirect is broken by GotoIf this info is changed.
+ that will causes confusion in dialplan execution.</p>
+<p>Resolves: #1273</p>
+<h4>res_musiconhold.c: Ensure we're always locked around music state access.</h4>
+<p>Author: Sean Bright
+ Date: 2025-04-08</p>
+<h4>res_musiconhold.c: Annotate when the channel is locked.</h4>
+<p>Author: Sean Bright
+ Date: 2025-04-08</p>
+<h4>res_musiconhold: Appropriately lock channel during start.</h4>
+<p>Author: Jaco Kroon
+ Date: 2024-12-19</p>
+<p>This relates to #829</p>
+<p>This doesn't sully solve the Ops issue, but it solves the specific crash
+ there. Further PRs to follow.</p>
+<p>In the specific crash the generator was still under construction when
+ moh was being stopped, which then proceeded to close the stream whilst
+ it was still in use.</p>
+<p>Signed-off-by: Jaco Kroon <a href="mailto:jaco@uls.co.za">jaco@uls.co.za</a></p>
+</body></html>
--- /dev/null
+
+## Change Log for Release asterisk-20.16.0-rc1
+
+### Links:
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0-rc1.html)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0-rc1)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0-rc1.tar.gz)
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
+
+### Summary:
+
+- Commits: 51
+- Commit Authors: 22
+- Issues Resolved: 37
+- Security Advisories Resolved: 0
+
+### User Notes:
+
+- #### app_queue.c: Add new global 'log_unpause_on_reason_change'
+ Add new global option 'log_unpause_on_reason_change' that
+ is default disabled. When enabled cause addition of UNPAUSE event on
+ every re-PAUSE with reason changed.
+
+- #### pbx_builtins: Allow custom tone for WaitExten.
+ The tone used while waiting for digits in WaitExten
+ can now be overridden by specifying an argument for the 'd'
+ option.
+
+- #### res_tonedetect: Add option for TONE_DETECT detection to auto stop.
+ The 'e' option for TONE_DETECT now allows detection to
+ be disabled automatically once the desired number of matches have
+ been fulfilled, which can help prevent race conditions in the
+ dialplan, since TONE_DETECT does not need to be disabled after
+ a hit.
+
+- #### sorcery: Prevent duplicate objects and ensure missing objects are created on u..
+ Users relying on Sorcery multiple writable backends configurations
+ (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
+ in sorcery.conf to ensure missing objects are recreated after temporary backend
+ failures. Default behavior remains unchanged unless explicitly enabled.
+
+- #### chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
+ A new WebSocket channel driver option `v` has been added to the
+ Dial application that allows you to specify additional URI parameters on
+ outgoing connections. Run `core show application Dial` from the Asterisk CLI
+ to see how to use it.
+
+- #### app_chanspy: Add option to not automatically answer channel.
+ ChanSpy and ExtenSpy can now be configured to not
+ automatically answer the channel by using the 'N' option.
+
+- #### cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
+ Enabling the tracking of the
+ STREAM_BEGIN and the STREAM_END event
+ types in cel.conf will log media files and
+ music on hold played to each channel.
+ The STREAM_BEGIN event's extra field will
+ contain a JSON with the file details (path,
+ format and language), or the class name, in
+ case of music on hold is played. The DTMF
+ event's extra field will contain a JSON with
+ the digit and the duration in milliseconds.
+
+- #### res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
+ Options are now available in the menuselect "Resource Modules"
+ category that allow you to enable the AES_192, AES_256 and AES_GCM
+ cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
+ them but modern versions do. Previously, the only way to enable them was
+ to set the CFLAGS environment variable when running ./configure.
+ The default setting is to disable them preserving existing behavior.
+
+- #### cdr: add CANCEL dispostion in CDR
+ A new CDR option "canceldispositionenabled" has been added
+ that when set to true, the NO ANSWER disposition will be split into
+ two dispositions: CANCEL and NO ANSWER. The default value is 'no'
+
+- #### func_curl: Allow auth methods to be set.
+ The httpauth field in CURLOPT now allows the authentication
+ methods to be set.
+
+- #### Media over Websocket Channel Driver
+ A new channel driver "chan_websocket" is now available. It can
+ exchange media over both inbound and outbound websockets and will both frame
+ and re-time the media it receives.
+ See http://s.asterisk.net/mow for more information.
+ The ARI channels/externalMedia API now includes support for the
+
+### Upgrade Notes:
+
+
+### Developer Notes:
+
+- #### ARI: Add command to indicate progress to a channel
+ A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel.
+
+- #### options: Change ast_options from ast_flags to ast_flags64.
+ The 32-bit ast_options has no room left to accomodate new
+ options and so has been converted to an ast_flags64 structure. All internal
+ references to ast_options have been updated to use the 64-bit flag
+ manipulation macros. External module references to the 32-bit ast_options
+ should continue to work on little-endian systems because the
+ least-significant bytes of a 64 bit integer will be in the same location as a
+ 32-bit integer. Because that's not the case on big-endian systems, we've
+ swapped the bytes in the flags manupulation macros on big-endian systems
+ so external modules should still work however you are encouraged to test.
+
+
+### Commit Authors:
+
+- Alexei Gradinari: (2)
+- Alexey Khabulyak: (2)
+- Allan Nathanson: (1)
+- Artem Umerov: (1)
+- Ben Ford: (1)
+- George Joseph: (9)
+- Igor Goncharovsky: (2)
+- Jaco Kroon: (1)
+- Joe Garlick: (1)
+- Jose Lopes: (1)
+- Kodokaii: (1)
+- Martin Tomec: (1)
+- Mike Bradeen: (1)
+- Mkmer: (1)
+- Naveen Albert: (15)
+- Sean Bright: (2)
+- Sperl Viktor: (2)
+- Stanislav Abramenkov: (1)
+- Stuart Henderson: (1)
+- Sven Kube: (2)
+- Tinet-Mucw: (2)
+- Zhou_jiajian: (1)
+
+## Issue and Commit Detail:
+
+### Closed Issues:
+
+ - 401: [bug]: app_dial: Answer Gosub option passthrough regression
+ - 927: [bug]: no audio when media source changed during the call
+ - 1176: [bug]: ast_slinear_saturated_multiply_float produces potentially audible distortion artifacts
+ - 1259: [bug]: New TenantID feature doesn't seem to set CDR for incoming calls
+ - 1260: [bug]: Asterisk sends RTP audio stream before ICE/DTLS completes
+ - 1269: [bug]: MixMonitor with D option produces corrupt file
+ - 1273: [bug]: When executed with GotoIf, the action Redirect does not take effect and causes confusion in dialplan execution.
+ - 1280: [improvement]: logging playback of audio per channel
+ - 1289: [bug]: sorcery - duplicate objects from multiple backends and backend divergence on update
+ - 1301: [bug]: sig_analog: fgccamamf doesn't handle STP, STP2, or STP3
+ - 1304: [bug]: FLUSH_MEDIA does not reset frame_queue_length in WebSocket channel
+ - 1305: [bug]: Realtime incorrectly falls back to next backend on record-not-found (SQL_NO_DATA), causing incorrect behavior and delay
+ - 1307: [improvement]: ast_tls_cert: Allow certificate validity to be configurable
+ - 1309: [bug]: Crash with C++ alternative storage backend enabled
+ - 1315: [bug]: When executed with dialplan, the action Redirect does not take effect.
+ - 1317: [bug]: AGI command buffer overflow with long variables
+ - 1321: [improvement]: app_agent_pool: Remove obsolete documentation
+ - 1323: [new-feature]: add CANCEL dispostion in CDR
+ - 1327: [bug]: res_stasis_device_state: can't delete ARI Devicestate after asterisk restart
+ - 1332: [new-feature]: func_curl: Allow auth methods to be set
+ - 1349: [bug]: Race condition on redirect can cause missing Diversion header
+ - 1352: [improvement]: Websocket channel with custom URI
+ - 1353: [bug]: AST_DATA_DIR/sounds/custom directory not searched
+ - 1358: [new-feature]: app_chanspy: Add option to not automatically answer channel
+ - 1364: [bug]: bridge.c: BRIDGE_NOANSWER not always obeyed
+ - 1366: [improvement]: func_frame_drop: Handle allocation failure properly
+ - 1369: [bug]: test_res_prometheus: Compilation failure in devmode due to curlopts not using long type
+ - 1371: [improvement]: func_frame_drop: Add debug messages for frames that can be dropped
+ - 1375: [improvement]: dsp.c: Improve logging in tone_detect().
+ - 1378: [bug]: chan_dahdi: dialmode feature is not properly reset between calls
+ - 1380: [bug]: sig_analog: Segfault due to calling strcmp on NULL
+ - 1384: [bug]: chan_websocket: asterisk crashes on hangup after STOP_MEDIA_BUFFERING command with id
+ - 1386: [bug]: enabling announceposition_only_up prevents any queue position announcements
+ - 1390: [improvement]: res_tonedetect: Add option to automatically end detection in TONE_DETECT
+ - 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
+ - 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable
+ - 1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage
+
+### Commits By Author:
+
+- #### Alexei Gradinari (2):
+ - res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)
+ - sorcery: Prevent duplicate objects and ensure missing objects are created on u..
+
+- #### Alexey Khabulyak (2):
+ - app_dial.c: Moved channel lock to prevent deadlock
+ - pbx_lua.c: segfault when pass null data to term_color function
+
+- #### Allan Nathanson (1):
+ - file.c: with "sounds_search_custom_dir = yes", search "custom" directory
+
+- #### Artem Umerov (1):
+ - Fix missing ast_test_flag64 in extconf.c
+
+- #### Ben Ford (1):
+ - res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
+
+- #### George Joseph (9):
+ - Media over Websocket Channel Driver
+ - app_mixmonitor: Update the documentation concerning the "D" option.
+ - cdr.c: Set tenantid from party_a->base instead of chan->base.
+ - options: Change ast_options from ast_flags to ast_flags64.
+ - res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
+ - channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.
+ - xmldoc.c: Fix rendering of CLI output.
+ - chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
+ - chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
+
+- #### Igor Goncharovsky (2):
+ - app_waitforsilence.c: Use milliseconds to calculate timeout time
+ - app_queue.c: Add new global 'log_unpause_on_reason_change'
+
+- #### Jaco Kroon (1):
+ - res_musiconhold: Appropriately lock channel during start.
+
+- #### Joe Garlick (1):
+ - chan_websocket.c: Add DTMF messages
+
+- #### Jose Lopes (1):
+ - res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
+
+- #### Martin Tomec (1):
+ - chan_pjsip.c: Change SSRC after media source change
+
+- #### Mike Bradeen (1):
+ - res_pjsip_diversion: resolve race condition between Diversion header processin..
+
+- #### Naveen Albert (15):
+ - sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.
+ - ast_tls_cert: Make certificate validity configurable.
+ - app_agent_pool: Remove documentation for removed option.
+ - func_curl: Allow auth methods to be set.
+ - bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.
+ - func_frame_drop: Handle allocation failure properly.
+ - test_res_prometheus: Fix compilation failure on Debian 13.
+ - func_frame_drop: Add debug messages for dropped frames.
+ - app_chanspy: Add option to not automatically answer channel.
+ - dsp.c: Improve debug logging in tone_detect().
+ - sig_analog: Fix SEGV due to calling strcmp on NULL.
+ - chan_dahdi: Fix erroneously persistent dialmode.
+ - sig_analog: Skip Caller ID spill if usecallerid=no.
+ - res_tonedetect: Add option for TONE_DETECT detection to auto stop.
+ - pbx_builtins: Allow custom tone for WaitExten.
+
+- #### Sean Bright (2):
+ - res_musiconhold.c: Annotate when the channel is locked.
+ - res_musiconhold.c: Ensure we're always locked around music state access.
+
+- #### Sperl Viktor (2):
+ - res_agi: Increase AGI command buffer size from 2K to 8K
+ - cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
+
+- #### Stanislav Abramenkov (1):
+ - bundled_pjproject: Avoid deadlock between transport and transaction
+
+- #### Stuart Henderson (1):
+ - app_queue: fix comparison for announce-position-only-up
+
+- #### Sven Kube (2):
+ - resource_channels.c: Don't call ast_channel_get_by_name on empty optional argu..
+ - ARI: Add command to indicate progress to a channel
+
+- #### Tinet-mucw (2):
+ - pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return..
+ - pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper sho..
+
+- #### kodokaii (1):
+ - chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
+
+- #### mkmer (1):
+ - utils.h: Add rounding to float conversion to int.
+
+- #### zhou_jiajian (1):
+ - cdr: add CANCEL dispostion in CDR
+
+
+### Commit List:
+
+- chan_websocket.c: Add DTMF messages
+- app_queue.c: Add new global 'log_unpause_on_reason_change'
+- app_waitforsilence.c: Use milliseconds to calculate timeout time
+- Fix missing ast_test_flag64 in extconf.c
+- pbx_builtins: Allow custom tone for WaitExten.
+- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
+- app_queue: fix comparison for announce-position-only-up
+- sig_analog: Skip Caller ID spill if usecallerid=no.
+- chan_dahdi: Fix erroneously persistent dialmode.
+- chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
+- sig_analog: Fix SEGV due to calling strcmp on NULL.
+- ARI: Add command to indicate progress to a channel
+- dsp.c: Improve debug logging in tone_detect().
+- res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
+- app_chanspy: Add option to not automatically answer channel.
+- xmldoc.c: Fix rendering of CLI output.
+- func_frame_drop: Add debug messages for dropped frames.
+- test_res_prometheus: Fix compilation failure on Debian 13.
+- func_frame_drop: Handle allocation failure properly.
+- pbx_lua.c: segfault when pass null data to term_color function
+- bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.
+- res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
+- app_dial.c: Moved channel lock to prevent deadlock
+- file.c: with "sounds_search_custom_dir = yes", search "custom" directory
+- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
+- channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.
+- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
+- cdr: add CANCEL dispostion in CDR
+- func_curl: Allow auth methods to be set.
+- options: Change ast_options from ast_flags to ast_flags64.
+- res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)
+- app_agent_pool: Remove documentation for removed option.
+- res_agi: Increase AGI command buffer size from 2K to 8K
+- ast_tls_cert: Make certificate validity configurable.
+- cdr.c: Set tenantid from party_a->base instead of chan->base.
+- app_mixmonitor: Update the documentation concerning the "D" option.
+- sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.
+- chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
+- chan_pjsip.c: Change SSRC after media source change
+- Media over Websocket Channel Driver
+- bundled_pjproject: Avoid deadlock between transport and transaction
+- utils.h: Add rounding to float conversion to int.
+- res_musiconhold.c: Ensure we're always locked around music state access.
+- res_musiconhold.c: Annotate when the channel is locked.
+- res_musiconhold: Appropriately lock channel during start.
+
+### Commit Details:
+
+#### chan_websocket.c: Add DTMF messages
+ Author: Joe Garlick
+ Date: 2025-09-04
+
+ Added DTMF messages to the chan_websocket feature.
+
+ When a user presses DTMF during a call over chan_websocket it will send a message like:
+ "DTMF_END digit:1"
+
+ Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70
+
+#### app_queue.c: Add new global 'log_unpause_on_reason_change'
+ Author: Igor Goncharovsky
+ Date: 2025-09-02
+
+ In many asterisk-based systems, the pause reason is used to separate
+ pauses by type,and logically, changing the reason defines two intervals
+ that should be accounted for separately. The introduction of a new
+ option allows me to separate the intervals of operator inactivity in
+ the log by the event of unpausing.
+
+ UserNote: Add new global option 'log_unpause_on_reason_change' that
+ is default disabled. When enabled cause addition of UNPAUSE event on
+ every re-PAUSE with reason changed.
+
+
+#### app_waitforsilence.c: Use milliseconds to calculate timeout time
+ Author: Igor Goncharovsky
+ Date: 2025-09-04
+
+ The functions WaitForNoise() and WaitForSilence() use the time()
+ functions to calculate elapsed time, which causes the timer to fire on
+ a whole second boundary, and the actual function execution time to fire
+ the timer may be 1 second less than expected. This fix replaces time()
+ with ast_tvnow().
+
+ Fixes: #1401
+
+#### Fix missing ast_test_flag64 in extconf.c
+ Author: Artem Umerov
+ Date: 2025-08-29
+
+ Fix missing ast_test_flag64 after https://github.com/asterisk/asterisk/commit/43bf8a4ded7a65203b766b91eaf8331a600e9d8d
+
+
+#### pbx_builtins: Allow custom tone for WaitExten.
+ Author: Naveen Albert
+ Date: 2025-08-25
+
+ Currently, the 'd' option will play dial tone while waiting
+ for digits. Allow it to accept an argument for any tone from
+ indications.conf.
+
+ Resolves: #1396
+
+ UserNote: The tone used while waiting for digits in WaitExten
+ can now be overridden by specifying an argument for the 'd'
+ option.
+
+
+#### res_tonedetect: Add option for TONE_DETECT detection to auto stop.
+ Author: Naveen Albert
+ Date: 2025-08-28
+
+ One of the problems with TONE_DETECT as it was originally written
+ is that if a tone is detected multiple times, it can trigger
+ the redirect logic multiple times as well. For example, if we
+ do an async goto in the dialplan after detecting a tone, because
+ the detector is still active until explicitly disabled, if we
+ detect the tone again, we will branch again and start executing
+ that dialplan a second time. This is rarely ever desired behavior,
+ and can happen if the detector is not removed quickly enough.
+
+ Add a new option, 'e', which automatically disables the detector
+ once the desired number of matches have been heard. This eliminates
+ the potential race condition where previously the detector would
+ need to be disabled immediately, but doing so quickly enough
+ was not guaranteed. This also allows match criteria to be retained
+ longer if needed, so the detector does not need to be destroyed
+ prematurely.
+
+ Resolves: #1390
+
+ UserNote: The 'e' option for TONE_DETECT now allows detection to
+ be disabled automatically once the desired number of matches have
+ been fulfilled, which can help prevent race conditions in the
+ dialplan, since TONE_DETECT does not need to be disabled after
+ a hit.
+
+
+#### app_queue: fix comparison for announce-position-only-up
+ Author: Stuart Henderson
+ Date: 2025-08-21
+
+ Numerically comparing that the current queue position is less than
+ last_pos_said can only be done after at least one announcement has been
+ made, otherwise last_pos_said is at the default (0).
+
+ Fixes: #1386
+
+#### sorcery: Prevent duplicate objects and ensure missing objects are created on u..
+ Author: Alexei Gradinari
+ Date: 2025-07-07
+
+ This patch resolves two issues in Sorcery objectset handling with multiple
+ backends:
+
+ 1. Prevent duplicate objects:
+ When an object exists in more than one backend (e.g., a contact in both
+ 'astdb' and 'realtime'), the objectset previously returned multiple instances
+ of the same logical object. This caused logic failures in components like the
+ PJSIP registrar, where duplicate contact entries led to overcounting and
+ incorrect deletions, when max_contacts=1 and remove_existing=yes.
+
+ This patch ensures only one instance of an object with a given key is added
+ to the objectset, avoiding these duplicate-related side effects.
+
+ 2. Ensure missing objects are created:
+ When using multiple writable backends, a temporary backend failure can lead
+ to objects missing permanently from that backend.
+ Currently, .update() silently fails if the object is not present,
+ and no .create() is attempted.
+ This results in inconsistent state across backends (e.g. astdb vs. realtime).
+
+ This patch introduces a new global option in sorcery.conf:
+ [general]
+ update_or_create_on_update_miss = yes|no
+
+ Default: no (preserves existing behavior).
+
+ When enabled: if .update() fails with no data found, .create() is attempted
+ in that backend. This ensures that objects missing due to temporary backend
+ outages are re-synchronized once the backend is available again.
+
+ Added a new CLI command:
+ sorcery show settings
+ Displays global Sorcery settings, including the current value of
+ update_or_create_on_update_miss.
+
+ Updated tests to validate both flag enabled/disabled behavior.
+
+ Fixes: #1289
+
+ UserNote: Users relying on Sorcery multiple writable backends configurations
+ (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
+ in sorcery.conf to ensure missing objects are recreated after temporary backend
+ failures. Default behavior remains unchanged unless explicitly enabled.
+
+
+#### sig_analog: Skip Caller ID spill if usecallerid=no.
+ Author: Naveen Albert
+ Date: 2025-08-25
+
+ If Caller ID is disabled for an FXS port, then we should not send any
+ Caller ID spill on the line, as we have no Caller ID information that
+ we can/should be sending.
+
+ Resolves: #1394
+
+#### chan_dahdi: Fix erroneously persistent dialmode.
+ Author: Naveen Albert
+ Date: 2025-08-18
+
+ It is possible to modify the dialmode setting in the chan_dahdi/sig_analog
+ private using the CHANNEL function, to modify it during calls. However,
+ it was not being reset between calls, meaning that if, for example, tone
+ dialing was disabled, it would never work again unless explicitly enabled.
+
+ This fixes the setting by pairing it with a "perm" version of the setting,
+ as a few other features have, so that it can be reset to the permanent
+ setting between calls. The documentation is also clarified to explain
+ the interaction of this setting and the digitdetect setting more clearly.
+
+ Resolves: #1378
+
+#### chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
+ Author: George Joseph
+ Date: 2025-08-13
+
+ * Added a new option to the WebSocket dial string to capture the additional
+ URI parameters.
+ * Added a new API ast_uri_verify_encoded() that verifies that a string
+ either doesn't need URI encoding or that it has already been encoded.
+ * Added a new API ast_websocket_client_add_uri_params() to add the params
+ to the client websocket session.
+ * Added XML documentation that will show up with `core show application Dial`
+ that shows how to use it.
+
+ Resolves: #1352
+
+ UserNote: A new WebSocket channel driver option `v` has been added to the
+ Dial application that allows you to specify additional URI parameters on
+ outgoing connections. Run `core show application Dial` from the Asterisk CLI
+ to see how to use it.
+
+
+#### chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
+ Author: George Joseph
+ Date: 2025-08-19
+
+ ast_websocket_read() receives data into a fixed 64K buffer then continually
+ reallocates a final buffer that, after all continuation frames have been
+ received, is the exact length of the data received and returns that to the
+ caller. process_text_message() in chan_websocket was attempting to set a
+ NULL terminator on the received payload assuming the payload buffer it
+ received was the large 64K buffer. The assumption was incorrect so when it
+ tried to set a NULL terminator on the payload, it could, depending on the
+ state of the heap at the time, cause heap corruption.
+
+ process_text_message() now allocates its own payload_len + 1 sized buffer,
+ copies the payload received from ast_websocket_read() into it then NULL
+ terminates it prevent the possibility of the overrun and corruption.
+
+ Resolves: #1384
+
+#### sig_analog: Fix SEGV due to calling strcmp on NULL.
+ Author: Naveen Albert
+ Date: 2025-08-18
+
+ Add an additional check to guard against the channel application being
+ NULL.
+
+ Resolves: #1380
+
+#### ARI: Add command to indicate progress to a channel
+ Author: Sven Kube
+ Date: 2025-07-30
+
+ Adds an ARI command to send a progress indication to a channel.
+
+ DeveloperNote: A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel.
+
+#### dsp.c: Improve debug logging in tone_detect().
+ Author: Naveen Albert
+ Date: 2025-08-15
+
+ The debug logging during DSP processing has always been kind
+ of overwhelming and annoying to troubleshoot. Simplify and
+ improve the logging in a few ways to aid DSP debugging:
+
+ * If we had a DSP hit, don't also emit the previous debug message that
+ was always logged. It is duplicated by the hit message, so this can
+ reduce the number of debug messages during detection by 50%.
+ * Include the hit count and required number of hits in the message so
+ on partial detections can be more easily troubleshot.
+ * Use debug level 9 for hits instead of 10, so we can focus on hits
+ without all the noise from the per-frame debug message.
+ * 1-index the hit count in the debug messages. On the first hit, it
+ currently logs '0', just as when we are not detecting anything,
+ which can be confusing.
+
+ Resolves: #1375
+
+#### res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
+ Author: Jose Lopes
+ Date: 2025-07-30
+
+ After an asterisk restart, the deletion of ARI Devicestates didn't
+ return error, but the devicestate was not deleted.
+ Found a typo on populate_cache function that created wrong cache for
+ device states.
+ This bug caused wrong assumption that devicestate didn't exist,
+ since it was not in cache, so deletion didn't returned error.
+
+ Fixes: #1327
+
+#### app_chanspy: Add option to not automatically answer channel.
+ Author: Naveen Albert
+ Date: 2025-08-13
+
+ Add an option for ChanSpy and ExtenSpy to not answer the channel
+ automatically. Most applications that auto-answer by default
+ already have an option to disable this behavior if unwanted.
+
+ Resolves: #1358
+
+ UserNote: ChanSpy and ExtenSpy can now be configured to not
+ automatically answer the channel by using the 'N' option.
+
+
+#### xmldoc.c: Fix rendering of CLI output.
+ Author: George Joseph
+ Date: 2025-08-14
+
+ If you do a `core show application Dial`, you'll see it's kind of a mess.
+ Indents are wrong is some places, examples are printed in black which makes
+ them invisible on most terminals, and the lack of line breaks in some cases
+ makes it hard to follow.
+
+ * Fixed the rendering of examples so they are indented properly and changed
+ the color so they can be seen.
+ * There is now a line break before each option.
+ * Options are now printed on their own line with all option content indented
+ below them.
+
+ Example from Dial before fixes:
+ ```
+ Example: Dial 555-1212 on first available channel in group 1, searching
+ from highest to lowest
+
+ Example: Ringing FXS channel 4 with ring cadence 2
+
+ Example: Dial 555-1212 on channel 3 and require answer confirmation
+
+ ...
+
+ O([mode]):
+ mode - With <mode> either not specified or set to '1', the originator
+ hanging up will cause the phone to ring back immediately.
+ - With <mode> set to '2', when the operator flashes the trunk, it will ring
+ their phone back.
+ Enables *operator services* mode. This option only works when bridging a DAHDI
+ channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it
+ will be ignored. When the destination answers (presumably an operator services
+ station), the originator no longer has control of their line. They may hang up,
+ but the switch will not release their line until the destination party (the
+ operator) hangs up.
+
+ p: This option enables screening mode. This is basically Privacy mode
+ without memory.
+ ```
+
+ After:
+ ```
+ Example: Dial 555-1212 on first available channel in group 1, searching
+ from highest to lowest
+
+ same => n,Dial(DAHDI/g1/5551212)
+
+ Example: Ringing FXS channel 4 with ring cadence 2
+
+ same => n,Dial(DAHDI/4r2)
+
+ Example: Dial 555-1212 on channel 3 and require answer confirmation
+
+ same => n,Dial(DAHDI/3c/5551212)
+
+ ...
+
+ O([mode]):
+ mode - With <mode> either not specified or set to '1', the originator
+ hanging up will cause the phone to ring back immediately.
+ With <mode> set to '2', when the operator flashes the trunk, it will
+ ring their phone back.
+ Enables *operator services* mode. This option only works when bridging
+ a DAHDI channel to another DAHDI channel only. If specified on
+ non-DAHDI interfaces, it will be ignored. When the destination answers
+ (presumably an operator services station), the originator no longer has
+ control of their line. They may hang up, but the switch will not
+ release their line until the destination party (the operator) hangs up.
+
+ p:
+ This option enables screening mode. This is basically Privacy mode
+ without memory.
+ ```
+
+ There are still things we can do to make this more readable but this is a
+ start.
+
+
+#### func_frame_drop: Add debug messages for dropped frames.
+ Author: Naveen Albert
+ Date: 2025-08-14
+
+ Add debug messages in scenarios where frames that are usually processed
+ are dropped or skipped.
+
+ Resolves: #1371
+
+#### test_res_prometheus: Fix compilation failure on Debian 13.
+ Author: Naveen Albert
+ Date: 2025-08-14
+
+ curl_easy_setopt expects long types, so be explicit.
+
+ Resolves: #1369
+
+#### func_frame_drop: Handle allocation failure properly.
+ Author: Naveen Albert
+ Date: 2025-08-14
+
+ Handle allocation failure and simplify the allocation using asprintf.
+
+ Resolves: #1366
+
+#### pbx_lua.c: segfault when pass null data to term_color function
+ Author: Alexey Khabulyak
+ Date: 2025-08-14
+
+ This can be reproduced under certain curcomstences.
+ For example: call app.playback from lua with invalid data: app.playback({}).
+ pbx_lua.c will try to get data for this playback using lua_tostring function.
+ This function returs NULL for everything but strings and numbers.
+ Then, it calls term_color with NULL data.
+ term_color function can call(if we don't use vt100 compat term)
+ ast_copy_string with NULL inbuf which cause segfault. bt example:
+ ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0)
+ at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412
+
+ Resolves: https://github.com/asterisk/asterisk/issues/1363
+
+#### bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.
+ Author: Naveen Albert
+ Date: 2025-08-14
+
+ If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed
+ to answer when another channel bridges to it using Bridge(), and this is
+ checked when ast_bridge_call* is called. However, another path exists
+ (bridge_exec -> ast_bridge_add_channel) where this variable was not
+ checked and channels would be answered. We now check the variable there.
+
+ Resolves: #401
+ Resolves: #1364
+
+#### res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
+ Author: Ben Ford
+ Date: 2025-08-04
+
+ There was no check in __rtp_sendto that prevented Asterisk from sending
+ RTP before DTLS had finished negotiating. This patch adds logic to do
+ so.
+
+ Fixes: #1260
+
+#### app_dial.c: Moved channel lock to prevent deadlock
+ Author: Alexey Khabulyak
+ Date: 2025-08-04
+
+ It's reproducible with pbx_lua, not regular dialplan.
+
+ deadlock description:
+ 1. asterisk locks a channel
+ 2. calls function onedigit_goto
+ 3. calls ast_goto_if_exists funciton
+ 4. checks ast_exists_extension -> pbx_extension_helper
+ 5. pbx_extension_helper calls pbx_find_extension
+ 6. Then asterisk starts autoservice in a new thread
+ 7. autoservice run tries to lock the channel again
+
+ Because our channel is locked already, autoservice can't lock.
+ Autoservice can't lock -> autoservice stop is waiting forever.
+ onedigit_goto waits for autoservice stop.
+
+ Resolves: https://github.com/asterisk/asterisk/issues/1335
+
+#### res_pjsip_diversion: resolve race condition between Diversion header processin..
+ Author: Mike Bradeen
+ Date: 2025-08-07
+
+ Based on the firing order of the PJSIP call-backs on a redirect, it was possible for
+ the Diversion header to not be included in the outgoing 181 response to the UAC and
+ the INVITE to the UAS.
+
+ This change moves the Diversion header processing to an earlier PJSIP callback while also
+ preventing the corresponding update that can cause a duplicate 181 response when processing
+ the header at that time.
+
+ Resolves: #1349
+
+#### file.c: with "sounds_search_custom_dir = yes", search "custom" directory
+ Author: Allan Nathanson
+ Date: 2025-08-10
+
+ With `sounds_search_custom_dir = yes`, we are supposed to search for sounds
+ in the `AST_DATA_DIR/sounds/custom` directory before searching the normal
+ directories. Unfortunately, a recent change
+ (https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in
+ the "custom" directory not being searched. This change restores this
+ expected behavior.
+
+ Resolves: #1353
+
+#### cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
+ Author: Sperl Viktor
+ Date: 2025-06-30
+
+ Fixes: #1280
+
+ UserNote: Enabling the tracking of the
+ STREAM_BEGIN and the STREAM_END event
+ types in cel.conf will log media files and
+ music on hold played to each channel.
+ The STREAM_BEGIN event's extra field will
+ contain a JSON with the file details (path,
+ format and language), or the class name, in
+ case of music on hold is played. The DTMF
+ event's extra field will contain a JSON with
+ the digit and the duration in milliseconds.
+
+
+#### channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.
+ Author: George Joseph
+ Date: 2025-07-30
+
+ The fact that deleting an object from a map invalidates any iterator
+ that happens to currently point to that object was overlooked in the initial
+ implementation. Unfortunately, there's no way to detect that an iterator
+ has been invalidated so the result was an occasional SEGV triggered by modules
+ like app_chanspy that opens an iterator and can keep it open for a long period
+ of time. The new implementation doesn't keep the underlying C++ iterator
+ open across calls to ast_channel_iterator_next() and uses a read lock
+ on the map to ensure that, even for the few microseconds we use the
+ iterator, another thread can't delete a channel from under it. Even with
+ this change, the iterators are still WAY faster than the ao2_legacy
+ storage driver.
+
+ Full details about the new implementation are located in the comments for
+ iterator_next() in channelstorage_cpp_map_name_id.cc.
+
+ Resolves: #1309
+
+#### res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
+ Author: George Joseph
+ Date: 2025-08-05
+
+ UserNote: Options are now available in the menuselect "Resource Modules"
+ category that allow you to enable the AES_192, AES_256 and AES_GCM
+ cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
+ them but modern versions do. Previously, the only way to enable them was
+ to set the CFLAGS environment variable when running ./configure.
+ The default setting is to disable them preserving existing behavior.
+
+
+#### cdr: add CANCEL dispostion in CDR
+ Author: zhou_jiajian
+ Date: 2025-07-24
+
+ In the original implementation, both CANCEL and NO ANSWER states were
+ consolidated under the NO ANSWER disposition. This patch introduces a
+ separate CANCEL disposition, with an optional configuration switch to
+ enable this new disposition.
+
+ Resolves: #1323
+
+ UserNote: A new CDR option "canceldispositionenabled" has been added
+ that when set to true, the NO ANSWER disposition will be split into
+ two dispositions: CANCEL and NO ANSWER. The default value is 'no'
+
+
+#### func_curl: Allow auth methods to be set.
+ Author: Naveen Albert
+ Date: 2025-08-01
+
+ Currently the CURL function only supports Basic Authentication,
+ the default auth method in libcurl. Add an option that also
+ allows enabling digest authentication.
+
+ Resolves: #1332
+
+ UserNote: The httpauth field in CURLOPT now allows the authentication
+ methods to be set.
+
+
+#### options: Change ast_options from ast_flags to ast_flags64.
+ Author: George Joseph
+ Date: 2025-07-21
+
+ DeveloperNote: The 32-bit ast_options has no room left to accomodate new
+ options and so has been converted to an ast_flags64 structure. All internal
+ references to ast_options have been updated to use the 64-bit flag
+ manipulation macros. External module references to the 32-bit ast_options
+ should continue to work on little-endian systems because the
+ least-significant bytes of a 64 bit integer will be in the same location as a
+ 32-bit integer. Because that's not the case on big-endian systems, we've
+ swapped the bytes in the flags manupulation macros on big-endian systems
+ so external modules should still work however you are encouraged to test.
+
+
+#### res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)
+ Author: Alexei Gradinari
+ Date: 2025-07-15
+
+ This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly
+ falls back to the next configured backend when the current one returns
+ SQL_NO_DATA (i.e., no record found).
+ This is a logical error and performance risk in multi-backend configurations.
+
+ Solution:
+ Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker.
+ ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found.
+ Core engine stops iterating on this marker, avoiding unnecessary fallback.
+
+ Notes:
+ Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly.
+ This patch only covers ODBC.
+
+ Fixes: #1305
+
+#### resource_channels.c: Don't call ast_channel_get_by_name on empty optional argu..
+ Author: Sven Kube
+ Date: 2025-07-30
+
+ `ast_ari_channels_create` and `ast_ari_channels_dial` called the
+ `ast_channel_get_by_name` function with optional arguments. Since
+ 8f1982c4d6, this function logs an error for empty channel names.
+ This commit adds checks for empty optional arguments that are used
+ to call `ast_channel_get_by_name` to prevent these error logs.
+
+
+#### app_agent_pool: Remove documentation for removed option.
+ Author: Naveen Albert
+ Date: 2025-07-28
+
+ The already-deprecated "password" option for the AGENT function was
+ removed in commit d43b17a872e8227aa8a9905a21f90bd48f9d5348 for
+ Asterisk 12, but the documentation for it wasn't removed then.
+
+ Resolves: #1321
+
+#### pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper sho..
+ Author: Tinet-mucw
+ Date: 2025-07-22
+
+ Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect.
+ In the situation that action Redirect is broken by pbx_extension_helper this info is changed.
+ This will cause the current dialplan location to be executed twice.
+ In other words, the Redirect action does not take effect.
+
+ Resolves: #1315
+
+#### res_agi: Increase AGI command buffer size from 2K to 8K
+ Author: Sperl Viktor
+ Date: 2025-07-22
+
+ Fixes: #1317
+
+#### ast_tls_cert: Make certificate validity configurable.
+ Author: Naveen Albert
+ Date: 2025-07-16
+
+ Currently, the ast_tls_cert script is hardcoded to produce certificates
+ with a validity of 365 days, which is not generally desirable for self-
+ signed certificates. Make this parameter configurable.
+
+ Resolves: #1307
+
+#### cdr.c: Set tenantid from party_a->base instead of chan->base.
+ Author: George Joseph
+ Date: 2025-07-17
+
+ The CDR tenantid was being set in cdr_object_alloc from the channel->base
+ snapshot. Since this happens at channel creation before the dialplan is even
+ reached, calls to `CHANNEL(tenantid)=<something>` in the dialplan were being
+ ignored. Instead we now take tenantid from party_a when
+ cdr_object_create_public_records() is called which is after the call has
+ ended and all channel snapshots rebuilt. This is exactly how accountcode
+ and amaflags, which can also be set in tha dialplpan, are handled.
+
+ Resolves: #1259
+
+#### app_mixmonitor: Update the documentation concerning the "D" option.
+ Author: George Joseph
+ Date: 2025-07-16
+
+ When using the "D" option to output interleaved audio, the file extension
+ must be ".raw". That info wasn't being properly rendered in the markdown
+ and HTML on the documentation site. The XML was updated to move the
+ note in the option section to a warning in the description.
+
+ Resolves: #1269
+
+#### sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.
+ Author: Naveen Albert
+ Date: 2025-07-14
+
+ Previously, we were only using # (ST) as a terminator, and not handling
+ A (STP), B (ST2P), or C (ST3P), which erroneously led to it being
+ treated as part of the dialed number. Parse any of these as the start
+ digit.
+
+ Resolves: #1301
+
+#### chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
+ Author: kodokaii
+ Date: 2025-07-03
+
+ In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from
+ the queue but does not reset the frame_queue_length counter.
+
+ As a result, the driver incorrectly thinks the queue is full after flushing,
+ which prevents new multimedia frames from being sent, especially after multiple
+ flush commands.
+
+ This fix sets frame_queue_length to 0 after flushing, ensuring the queue state
+ is consistent with its actual content.
+
+ Fixes: #1304
+
+#### chan_pjsip.c: Change SSRC after media source change
+ Author: Martin Tomec
+ Date: 2025-06-25
+
+ When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly.
+ To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream.
+ This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.
+
+ Resolves: #927
+
+#### Media over Websocket Channel Driver
+ Author: George Joseph
+ Date: 2025-04-28
+
+ * Created chan_websocket which can exchange media over both inbound and
+ outbound websockets which the driver will frame and time.
+ See http://s.asterisk.net/mow for more information.
+
+ * res_http_websocket: Made defines for max message size public and converted
+ a few nuisance verbose messages to debugs.
+
+ * main/channel.c: Changed an obsolete nuisance error to a debug.
+
+ * ARI channels: Updated externalMedia to include chan_websocket as a supported
+ transport.
+
+ UserNote: A new channel driver "chan_websocket" is now available. It can
+ exchange media over both inbound and outbound websockets and will both frame
+ and re-time the media it receives.
+ See http://s.asterisk.net/mow for more information.
+
+ UserNote: The ARI channels/externalMedia API now includes support for the
+ WebSocket transport provided by chan_websocket.
+
+
+#### bundled_pjproject: Avoid deadlock between transport and transaction
+ Author: Stanislav Abramenkov
+ Date: 2025-07-01
+
+ Backport patch from upstream
+ * Avoid deadlock between transport and transaction
+ https://github.com/pjsip/pjproject/commit/edde06f261ac
+
+ Issue described in
+ https://github.com/pjsip/pjproject/issues/4442
+
+
+#### utils.h: Add rounding to float conversion to int.
+ Author: mkmer
+ Date: 2025-03-23
+
+ Quote from an audio engineer NR9V:
+ There is a minor issue of a small amount of crossover distortion though as a result of `ast_slinear_saturated_multiply_float()` not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion.
+
+ Resolves: #1176
+
+#### pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return..
+ Author: Tinet-mucw
+ Date: 2025-06-18
+
+ Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect.
+ In the situation that action Redirect is broken by GotoIf this info is changed.
+ that will causes confusion in dialplan execution.
+
+ Resolves: #1273
+
+#### res_musiconhold.c: Ensure we're always locked around music state access.
+ Author: Sean Bright
+ Date: 2025-04-08
+
+
+#### res_musiconhold.c: Annotate when the channel is locked.
+ Author: Sean Bright
+ Date: 2025-04-08
+
+
+#### res_musiconhold: Appropriately lock channel during start.
+ Author: Jaco Kroon
+ Date: 2024-12-19
+
+ This relates to #829
+
+ This doesn't sully solve the Ops issue, but it solves the specific crash
+ there. Further PRs to follow.
+
+ In the specific crash the generator was still under construction when
+ moh was being stopped, which then proceeded to close the stream whilst
+ it was still in use.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
projects / thirdparty / asterisk.git / commitdiff
