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Update for 21.9.0-rc1

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<html><head><title>ChangeLog for asterisk-21.9.0-rc1</title></head><body>
<h2>Change Log for Release asterisk-21.9.0-rc1</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0-rc1.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0-rc1">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0-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: 24</li>
<li>Commit Authors: 18</li>
<li>Issues Resolved: 12</li>
<li>Security Advisories Resolved: 0</li>
</ul>
<h3>User Notes:</h3>
<ul>
<li>
<h4>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</h4>
<p>A Dial timeout on POST /channels/{channelId}/dial will now result in a
CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.</p>
</li>
<li>
<h4>contrib: Add systemd service and timer files for malloc trim.</h4>
<p>Service and timer files for systemd have been added to the
contrib/systemd/ directory. If you are experiencing memory issues,
install these files to have "malloc trim" periodically run on the
system.</p>
</li>
<li>
<h4>Add log-caller-id-name option to log Caller ID Name in queue log</h4>
<p>This patch adds a global configuration option, log-caller-id-name, to queues.conf
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
Any '|' characters in the caller ID name will be replaced with '_'.
(provided it’s allowed by the existing log_restricted_caller_id rules).
When log-caller-id-name=no (the default), the Caller ID name is omitted.</p>
</li>
<li>
<h4>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</h4>
<p>In cli.conf, you can now define startup commands that run before
core initialization and before module initialization.</p>
</li>
<li>
<h4>audiosocket: added support for DTMF frames</h4>
<p>The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).</p>
</li>
</ul>
<h3>Upgrade Notes:</h3>
<ul>
<li>
<h4>ARI: REST over Websocket</h4>
This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</li>
</ul>
<h3>Commit Authors:</h3>
<ul>
<li>Albrecht Oster: (1)</li>
<li>Alexei Gradinari: (1)</li>
<li>Allan Nathanson: (1)</li>
<li>Andreas Wehrmann: (1)</li>
<li>Ben Ford: (1)</li>
<li>Florent CHAUVEAU: (1)</li>
<li>George Joseph: (4)</li>
<li>Joshua C. Colp: (1)</li>
<li>Luz Paz: (1)</li>
<li>Mark Murawski: (1)</li>
<li>Mike Bradeen: (1)</li>
<li>Mkmer: (1)</li>
<li>Naveen Albert: (3)</li>
<li>Norm Harrison: (2)</li>
<li>Peter Jannesen: (1)</li>
<li>Phoneben: (1)</li>
<li>Sean Bright: (1)</li>
<li>Zhai Liangliang: (1)</li>
</ul>
<h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3>
<ul>
<li>505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()</li>
<li>643: [new-feature]: pjsip show contact -- show all details same as AMI PJSIPShowContacts</li>
<li>963: [bug]: missing hangup cause for ARI ChannelDestroyed when Dial times out</li>
<li>1091: [improvement]: app queue :add to queue log callerid name</li>
<li>1144: [bug]: action_redirect don't remove bridge_after_goto data</li>
<li>1171: [improvement]: Need the capability in audiohook.c for fractional (float) type volume adjustments.</li>
<li>1181: [bug]: Incorrect PJSIP Endpoint Device States on Multiple Channels</li>
<li>1190: [bug]: Crash when starting ConfBridge recording over CLI and AMI</li>
<li>1197: [bug]: ChannelHangupRequest does not show cause code in all cases</li>
<li>1206: [improvement]: chan_iax2: Minor improvements to documentation and warning messages.</li>
<li>1220: [bug]: res_pjsip_caller_id: OLI is not parsed if contained in a URI parameter</li>
<li>1224: [improvement]: app_meetme: Removal version is incorrect</li>
</ul>
<h3>Commits By Author:</h3>
<ul>
<li>
<h4>Albrecht Oster (1):</h4>
</li>
<li>
<p>res_pjproject: Fix DTLS client check failing on some platforms</p>
</li>
<li>
<h4>Alexei Gradinari (1):</h4>
</li>
<li>
<p>chan_pjsip: set correct Endpoint Device State on multiple channels</p>
</li>
<li>
<h4>Allan Nathanson (1):</h4>
</li>
<li>
<p>file.c: missing "custom" sound files should not generate warning logs</p>
</li>
<li>
<h4>Andreas Wehrmann (1):</h4>
</li>
<li>
<p>pbx_ael: unregister AELSub application and CLI commands on module load failure</p>
</li>
<li>
<h4>Ben Ford (1):</h4>
</li>
<li>
<p>contrib: Add systemd service and timer files for malloc trim.</p>
</li>
<li>
<h4>Florent CHAUVEAU (1):</h4>
</li>
<li>
<p>audiosocket: added support for DTMF frames</p>
</li>
<li>
<h4>George Joseph (4):</h4>
</li>
<li>ARI: REST over Websocket</li>
<li>ari_websockets: Fix frack if ARI config fails to load.</li>
<li>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</li>
<li>
<p>Prequisites for ARI Outbound Websockets</p>
</li>
<li>
<h4>Joshua C. Colp (1):</h4>
</li>
<li>
<p>channel: Always provide cause code in ChannelHangupRequest.</p>
</li>
<li>
<h4>Luz Paz (1):</h4>
</li>
<li>
<p>docs: Fix typos in apps/</p>
</li>
<li>
<h4>Mark Murawski (1):</h4>
</li>
<li>
<p>chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..</p>
</li>
<li>
<h4>Mike Bradeen (1):</h4>
</li>
<li>
<p>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</p>
</li>
<li>
<h4>Naveen Albert (3):</h4>
</li>
<li>chan_iax2: Minor improvements to documentation and warning messages.</li>
<li>app_meetme: Remove inaccurate removal version from xmldocs.</li>
<li>
<p>res_pjsip_caller_id: Also parse URI parameters for ANI2.</p>
</li>
<li>
<h4>Norm Harrison (2):</h4>
</li>
<li>audiosocket: fix timeout, fix dialplan app exit, server address in logs</li>
<li>
<p>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</p>
</li>
<li>
<h4>Peter Jannesen (1):</h4>
</li>
<li>
<p>action_redirect: remove after_bridge_goto_info</p>
</li>
<li>
<h4>Sean Bright (1):</h4>
</li>
<li>
<p>app_confbridge: Prevent crash when publishing channel-less event.</p>
</li>
<li>
<h4>Zhai Liangliang (1):</h4>
</li>
<li>
<p>Update config.guess and config.sub</p>
</li>
<li>
<h4>mkmer (1):</h4>
</li>
<li>
<p>audiohook.c: Add ability to adjust volume with float</p>
</li>
<li>
<h4>phoneben (1):</h4>
</li>
<li>Add log-caller-id-name option to log Caller ID Name in queue log</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>res_pjsip_caller_id: Also parse URI parameters for ANI2.</li>
<li>app_meetme: Remove inaccurate removal version from xmldocs.</li>
<li>docs: Fix typos in apps/</li>
<li>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</li>
<li>chan_iax2: Minor improvements to documentation and warning messages.</li>
<li>pbx_ael: unregister AELSub application and CLI commands on module load failure</li>
<li>res_pjproject: Fix DTLS client check failing on some platforms</li>
<li>Prequisites for ARI Outbound Websockets</li>
<li>contrib: Add systemd service and timer files for malloc trim.</li>
<li>action_redirect: remove after_bridge_goto_info</li>
<li>channel: Always provide cause code in ChannelHangupRequest.</li>
<li>Add log-caller-id-name option to log Caller ID Name in queue log</li>
<li>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</li>
<li>app_confbridge: Prevent crash when publishing channel-less event.</li>
<li>ari_websockets: Fix frack if ARI config fails to load.</li>
<li>ARI: REST over Websocket</li>
<li>audiohook.c: Add ability to adjust volume with float</li>
<li>audiosocket: added support for DTMF frames</li>
<li>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</li>
<li>audiosocket: fix timeout, fix dialplan app exit, server address in logs</li>
<li>Update config.guess and config.sub</li>
<li>chan_pjsip: set correct Endpoint Device State on multiple channels</li>
<li>file.c: missing "custom" sound files should not generate warning logs</li>
</ul>
<h3>Commit Details:</h3>
<h4>res_pjsip_caller_id: Also parse URI parameters for ANI2.</h4>
<p>Author: Naveen Albert
Date: 2025-04-26</p>
<p>If the isup-oli was sent as a URI parameter, rather than a header
parameter, it was not being parsed. Make sure we parse both if
needed so the ANI2 is set regardless of which type of parameter
the isup-oli is sent as.</p>
<p>Resolves: #1220</p>
<h4>app_meetme: Remove inaccurate removal version from xmldocs.</h4>
<p>Author: Naveen Albert
Date: 2025-04-26</p>
<p>app_meetme is deprecated but wasn't removed as planned in 21,
so remove the inaccurate removal version.</p>
<p>Resolves: #1224</p>
<h4>docs: Fix typos in apps/</h4>
<p>Author: Luz Paz
Date: 2025-04-09</p>
<p>Found via codespell</p>
<h4>stasis/control.c: Set Hangup Cause to No Answer on Dial timeout</h4>
<p>Author: Mike Bradeen
Date: 2025-04-17</p>
<p>Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs,
but the Dial command via ARI did not set an explicit reason. This resulted in a
CANCEL with Normal Call Clearing and corresponding ChannelDestroyed.</p>
<p>This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the
other operations.</p>
<p>Fixes: #963</p>
<p>UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a
CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.</p>
<h4>chan_iax2: Minor improvements to documentation and warning messages.</h4>
<p>Author: Naveen Albert
Date: 2025-04-18</p>
<ul>
<li>Update Dial() documentation for IAX2 to include syntax for RSA
public key names.</li>
<li>Add additional details to a couple warnings to provide more context
when an undecodable frame is received.</li>
</ul>
<p>Resolves: #1206</p>
<h4>pbx_ael: unregister AELSub application and CLI commands on module load failure</h4>
<p>Author: Andreas Wehrmann
Date: 2025-04-18</p>
<p>This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out,
that the AEL module doesn't do proper cleanup when it fails to load.
This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module()
returns an error but load_module() doesn't then unregister CLI cmds and the application.</p>
<h4>res_pjproject: Fix DTLS client check failing on some platforms</h4>
<p>Author: Albrecht Oster
Date: 2025-04-10</p>
<p>Certain platforms (mainly BSD derivatives) have an additional length
field in <code>sockaddr_in6</code> and <code>sockaddr_in</code>.
<code>ast_sockaddr_from_pj_sockaddr()</code> does not take this field into account
when copying over values from the <code>pj_sockaddr</code> into the <code>ast_sockaddr</code>.
The resulting <code>ast_sockaddr</code> will have an uninitialized value for
<code>sin6_len</code>/<code>sin_len</code> while the other <code>ast_sockaddr</code> (not converted from
a <code>pj_sockaddr</code>) to check against in <code>ast_sockaddr_pj_sockaddr_cmp()</code>
has the correct length value set.</p>
<p>This has the effect that <code>ast_sockaddr_cmp()</code> will always indicate
an address mismatch, because it does a bitwise comparison, and all DTLS
packets are dropped even if addresses and ports match.</p>
<p><code>ast_sockaddr_from_pj_sockaddr()</code> now checks whether the length fields
are available on the current platform and sets the values accordingly.</p>
<p>Resolves: #505</p>
<h4>Prequisites for ARI Outbound Websockets</h4>
<p>Author: George Joseph
Date: 2025-04-16</p>
<p>stasis:
* Added stasis_app_is_registered().
* Added stasis_app_control_mark_failed().
* Added stasis_app_control_is_failed().
* Fixed res_stasis_device_state so unsubscribe all works properly.
* Modified stasis_app_unregister() to unsubscribe from all event sources.
* Modified stasis_app_exec to return -1 if stasis_app_control_is_failed()
returns true.</p>
<p>http:
* Added ast_http_create_basic_auth_header().</p>
<p>md5:
* Added define for MD5_DIGEST_LENGTH.</p>
<p>tcptls:
* Added flag to ast_tcptls_session_args to suppress connection log messages
to give callers more control over logging.</p>
<p>http_websocket:
* Add flag to ast_websocket_client_options to suppress connection log messages
to give callers more control over logging.
* Added username and password to ast_websocket_client_options to support
outbound basic authentication.
* Added ast_websocket_result_to_str().</p>
<h4>contrib: Add systemd service and timer files for malloc trim.</h4>
<p>Author: Ben Ford
Date: 2025-04-16</p>
<p>Adds two files to the contrib/systemd/ directory that can be installed
to periodically run "malloc trim" on Asterisk. These files do nothing
unless they are explicitly moved to the correct location on the system.
Users who are experiencing Asterisk memory issues can use this service
to potentially help combat the problem. These files can also be
configured to change the start time and interval. See systemd.timer(5)
and systemd.time(7) for more information.</p>
<p>UserNote: Service and timer files for systemd have been added to the
contrib/systemd/ directory. If you are experiencing memory issues,
install these files to have "malloc trim" periodically run on the
system.</p>
<h4>action_redirect: remove after_bridge_goto_info</h4>
<p>Author: Peter Jannesen
Date: 2025-03-13</p>
<p>Under certain circumstances the context/extens/prio are stored in the
after_bridge_goto_info. This info is used when the bridge is broken by
for hangup of the other party. In the situation that the bridge is
broken by an AMI Redirect this info is not used but also not removed.
With the result that when the channel is put back in a bridge and the
bridge is broken the execution continues at the wrong
context/extens/prio.</p>
<p>Resolves: #1144</p>
<h4>channel: Always provide cause code in ChannelHangupRequest.</h4>
<p>Author: Joshua C. Colp
Date: 2025-04-16</p>
<p>When queueing a channel to be hung up a cause code can be
specified in one of two ways:</p>
<ol>
<li>
<p>ast_queue_hangup_with_cause
This function takes in a cause code and queues it as part
of the hangup request, which ultimately results in it being
set on the channel.</p>
</li>
<li>
<p>ast_channel_hangupcause_set + ast_queue_hangup
This combination sets the hangup cause on the channel before
queueing the hangup instead of as part of that process.</p>
</li>
</ol>
<p>In the #2 case the ChannelHangupRequest event would not contain
the cause code. For consistency if a cause code has been set
on the channel it will now be added to the event.</p>
<p>Resolves: #1197</p>
<h4>Add log-caller-id-name option to log Caller ID Name in queue log</h4>
<p>Author: phoneben
Date: 2025-02-28</p>
<p>Add log-caller-id-name option to log Caller ID Name in queue log</p>
<p>This patch introduces a new global configuration option, log-caller-id-name,
to queues.conf to control whether the Caller ID name is logged when a call enters a queue.</p>
<p>When log-caller-id-name=yes, the Caller ID name is logged
as parameter 4 in the queue log, provided it’s allowed by the
existing log_restricted_caller_id rules. If log-caller-id-name=no (the default),
the Caller ID name is omitted from the logs.</p>
<p>Fixes: #1091</p>
<p>UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
Any '|' characters in the caller ID name will be replaced with '_'.
(provided it’s allowed by the existing log_restricted_caller_id rules).
When log-caller-id-name=no (the default), the Caller ID name is omitted.</p>
<h4>asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.</h4>
<p>Author: George Joseph
Date: 2025-04-10</p>
<p>Commands in the "[startup_commands]" section of cli.conf have historically run
after all core and module initialization has been completed and just before
"Asterisk Ready" is printed on the console. This meant that if you
wanted to debug initialization of a specific module, your only option
was to turn on debug for everything by setting "debug" in asterisk.conf.</p>
<p>This commit introduces options to allow you to run CLI commands earlier in
the asterisk startup process.</p>
<p>A command with a value of "pre-init" will run just after logger initialization
but before most core, and all module, initialization.</p>
<p>A command with a value of "pre-module" will run just after all core
initialization but before all module initialization.</p>
<p>A command with a value of "fully-booted" (or "yes" for backwards
compatibility) will run as they always have been...after all
initialization and just before "Asterisk Ready" is printed on the console.</p>
<p>This means you could do this...</p>
<p><code>[startup_commands]
core set debug 3 res_pjsip.so = pre-module
core set debug 0 res_pjsip.so = fully-booted</code></p>
<p>This would turn debugging on for res_pjsip.so to catch any module
initialization debug messages then turn it off again after the module is
loaded.</p>
<p>UserNote: In cli.conf, you can now define startup commands that run before
core initialization and before module initialization.</p>
<h4>app_confbridge: Prevent crash when publishing channel-less event.</h4>
<p>Author: Sean Bright
Date: 2025-04-07</p>
<p>Resolves: #1190</p>
<h4>ari_websockets: Fix frack if ARI config fails to load.</h4>
<p>Author: George Joseph
Date: 2025-04-02</p>
<p>ari_ws_session_registry_dtor() wasn't checking that the container was valid
before running ao2_callback on it to shutdown registered sessions.</p>
<h4>ARI: REST over Websocket</h4>
<p>Author: George Joseph
Date: 2025-03-12</p>
<p>This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.</p>
<p>For full details on how to use the new capability, visit...</p>
<p>https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</p>
<p>Changes:</p>
<ul>
<li>Added utilities to http.c:<ul>
<li>ast_get_http_method_from_string().</li>
<li>ast_http_parse_post_form().</li>
</ul>
</li>
<li>Added utilities to json.c:<ul>
<li>ast_json_nvp_array_to_ast_variables().</li>
<li>ast_variables_to_json_nvp_array().</li>
</ul>
</li>
<li>Added definitions for new events to carry REST responses.</li>
<li>Created res/ari/ari_websocket_requests.c to house the new request handlers.</li>
<li>Moved non-event specific code out of res/ari/resource_events.c into
res/ari/ari_websockets.c</li>
<li>Refactored res/res_ari.c to move non-http code out of ast_ari_callback()
(which is http specific) and into ast_ari_invoke() so it can be shared
between both the http and websocket transports.</li>
</ul>
<p>UpgradeNote: This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/</p>
<h4>audiohook.c: Add ability to adjust volume with float</h4>
<p>Author: mkmer
Date: 2025-03-18</p>
<p>Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41.</p>
<p>This is accomplished by the following:
Convert internal variables to type float.
Always use ast_frame_adjust_volume_float() for adjustments.
Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust().
Cast float to int in ast_audiohook_volume_get()
Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float.</p>
<p>This update maintains 100% backward compatibility.</p>
<p>Resolves: #1171</p>
<h4>audiosocket: added support for DTMF frames</h4>
<p>Author: Florent CHAUVEAU
Date: 2025-02-28</p>
<p>Updated the AudioSocket protocol to allow sending DTMF frames.
AST_FRAME_DTMF frames are now forwarded to the server, in addition to
AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF
with value 0x03 was added to the protocol. The payload is a 1-byte
ascii representing the DTMF digit (0-9,*,#...).</p>
<p>UserNote: The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).</p>
<h4>asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'</h4>
<p>Author: Norm Harrison
Date: 2023-04-03</p>
<p>Co-authored-by: Florent CHAUVEAU <a href="&#109;&#97;&#105;&#108;&#116;&#111;&#58;&#102;&#108;&#111;&#114;&#101;&#110;&#116;&#99;&#104;&#64;&#112;&#109;&#46;&#109;&#101;">&#102;&#108;&#111;&#114;&#101;&#110;&#116;&#99;&#104;&#64;&#112;&#109;&#46;&#109;&#101;</a></p>
<h4>audiosocket: fix timeout, fix dialplan app exit, server address in logs</h4>
<p>Author: Norm Harrison
Date: 2023-04-03</p>
<ul>
<li>Correct wait timeout logic in the dialplan application.</li>
<li>Include server address in log messages for better traceability.</li>
<li>Allow dialplan app to exit gracefully on hangup messages and socket closure.</li>
<li>Optimize I/O by reducing redundant read()/write() operations.</li>
</ul>
<p>Co-authored-by: Florent CHAUVEAU <a href="&#109;&#97;&#105;&#108;&#116;&#111;&#58;&#102;&#108;&#111;&#114;&#101;&#110;&#116;&#99;&#104;&#64;&#112;&#109;&#46;&#109;&#101;">&#102;&#108;&#111;&#114;&#101;&#110;&#116;&#99;&#104;&#64;&#112;&#109;&#46;&#109;&#101;</a></p>
<h4>chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..</h4>
<p>Author: Mark Murawski
Date: 2025-03-23</p>
<p>CLI 'pjsip show contact' does not show enough information.
One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact
This feature adds the same details as PJSIPShowContacts to the CLI</p>
<p>Resolves: #643</p>
<h4>Update config.guess and config.sub</h4>
<p>Author: Zhai Liangliang
Date: 2025-03-26</p>
<h4>chan_pjsip: set correct Endpoint Device State on multiple channels</h4>
<p>Author: Alexei Gradinari
Date: 2025-03-25</p>
<ol>
<li>
<p>When one channel is placed on hold, the device state is set to ONHOLD
without checking other channels states.
In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN
to calculate aggregate device state of all active channels.</p>
</li>
<li>
<p>The current implementation incorrectly classifies channels in use.
The only channels that has the states: UP, RING and BUSY are considered as "in use".
A channel should be considered "in use" if its state is anything other than
DOWN or RESERVED.</p>
</li>
<li>
<p>Currently, if the number of channels "in use" is greater than device_state_busy_at,
the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate
device state.
The endpoint device state should be BUSY if the number of channels "in use" is greater
than or equal to device_state_busy_at.</p>
</li>
</ol>
<p>Fixes: #1181</p>
<h4>file.c: missing "custom" sound files should not generate warning logs</h4>
<p>Author: Allan Nathanson
Date: 2025-03-18</p>
<p>With <code>sounds_search_custom_dir = yes</code> we first look to see if a sound file
is present in the "custom" sound directory before looking in the standard
sound directories. We should not be issuing a WARNING log message if a
sound cannot be found in the "custom" directory.</p>
<p>Resolves: https://github.com/asterisk/asterisk/issues/1170</p>
</body></html>

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## Change Log for Release asterisk-21.9.0-rc1
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0-rc1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0-rc1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0-rc1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary:
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
### User Notes:
- #### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
A Dial timeout on POST /channels/{channelId}/dial will now result in a
CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
- #### contrib: Add systemd service and timer files for malloc trim.
Service and timer files for systemd have been added to the
contrib/systemd/ directory. If you are experiencing memory issues,
install these files to have "malloc trim" periodically run on the
system.
- #### Add log-caller-id-name option to log Caller ID Name in queue log
This patch adds a global configuration option, log-caller-id-name, to queues.conf
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
Any '|' characters in the caller ID name will be replaced with '_'.
(provided it’s allowed by the existing log_restricted_caller_id rules).
When log-caller-id-name=no (the default), the Caller ID name is omitted.
- #### asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
In cli.conf, you can now define startup commands that run before
core initialization and before module initialization.
- #### audiosocket: added support for DTMF frames
The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).
### Upgrade Notes:
- #### ARI: REST over Websocket
This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
### Commit Authors:
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)
## Issue and Commit Detail:
### Closed Issues:
- 505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
- 643: [new-feature]: pjsip show contact -- show all details same as AMI PJSIPShowContacts
- 963: [bug]: missing hangup cause for ARI ChannelDestroyed when Dial times out
- 1091: [improvement]: app queue :add to queue log callerid name
- 1144: [bug]: action_redirect don't remove bridge_after_goto data
- 1171: [improvement]: Need the capability in audiohook.c for fractional (float) type volume adjustments.
- 1181: [bug]: Incorrect PJSIP Endpoint Device States on Multiple Channels
- 1190: [bug]: Crash when starting ConfBridge recording over CLI and AMI
- 1197: [bug]: ChannelHangupRequest does not show cause code in all cases
- 1206: [improvement]: chan_iax2: Minor improvements to documentation and warning messages.
- 1220: [bug]: res_pjsip_caller_id: OLI is not parsed if contained in a URI parameter
- 1224: [improvement]: app_meetme: Removal version is incorrect
### Commits By Author:
- #### Albrecht Oster (1):
- res_pjproject: Fix DTLS client check failing on some platforms
- #### Alexei Gradinari (1):
- chan_pjsip: set correct Endpoint Device State on multiple channels
- #### Allan Nathanson (1):
- file.c: missing "custom" sound files should not generate warning logs
- #### Andreas Wehrmann (1):
- pbx_ael: unregister AELSub application and CLI commands on module load failure
- #### Ben Ford (1):
- contrib: Add systemd service and timer files for malloc trim.
- #### Florent CHAUVEAU (1):
- audiosocket: added support for DTMF frames
- #### George Joseph (4):
- ARI: REST over Websocket
- ari_websockets: Fix frack if ARI config fails to load.
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
- Prequisites for ARI Outbound Websockets
- #### Joshua C. Colp (1):
- channel: Always provide cause code in ChannelHangupRequest.
- #### Luz Paz (1):
- docs: Fix typos in apps/
- #### Mark Murawski (1):
- chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..
- #### Mike Bradeen (1):
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
- #### Naveen Albert (3):
- chan_iax2: Minor improvements to documentation and warning messages.
- app_meetme: Remove inaccurate removal version from xmldocs.
- res_pjsip_caller_id: Also parse URI parameters for ANI2.
- #### Norm Harrison (2):
- audiosocket: fix timeout, fix dialplan app exit, server address in logs
- asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'
- #### Peter Jannesen (1):
- action_redirect: remove after_bridge_goto_info
- #### Sean Bright (1):
- app_confbridge: Prevent crash when publishing channel-less event.
- #### Zhai Liangliang (1):
- Update config.guess and config.sub
- #### mkmer (1):
- audiohook.c: Add ability to adjust volume with float
- #### phoneben (1):
- Add log-caller-id-name option to log Caller ID Name in queue log
### Commit List:
- res_pjsip_caller_id: Also parse URI parameters for ANI2.
- app_meetme: Remove inaccurate removal version from xmldocs.
- docs: Fix typos in apps/
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
- chan_iax2: Minor improvements to documentation and warning messages.
- pbx_ael: unregister AELSub application and CLI commands on module load failure
- res_pjproject: Fix DTLS client check failing on some platforms
- Prequisites for ARI Outbound Websockets
- contrib: Add systemd service and timer files for malloc trim.
- action_redirect: remove after_bridge_goto_info
- channel: Always provide cause code in ChannelHangupRequest.
- Add log-caller-id-name option to log Caller ID Name in queue log
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
- app_confbridge: Prevent crash when publishing channel-less event.
- ari_websockets: Fix frack if ARI config fails to load.
- ARI: REST over Websocket
- audiohook.c: Add ability to adjust volume with float
- audiosocket: added support for DTMF frames
- asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'
- audiosocket: fix timeout, fix dialplan app exit, server address in logs
- Update config.guess and config.sub
- chan_pjsip: set correct Endpoint Device State on multiple channels
- file.c: missing "custom" sound files should not generate warning logs
### Commit Details:
#### res_pjsip_caller_id: Also parse URI parameters for ANI2.
Author: Naveen Albert
Date: 2025-04-26
If the isup-oli was sent as a URI parameter, rather than a header
parameter, it was not being parsed. Make sure we parse both if
needed so the ANI2 is set regardless of which type of parameter
the isup-oli is sent as.
Resolves: #1220
#### app_meetme: Remove inaccurate removal version from xmldocs.
Author: Naveen Albert
Date: 2025-04-26
app_meetme is deprecated but wasn't removed as planned in 21,
so remove the inaccurate removal version.
Resolves: #1224
#### docs: Fix typos in apps/
Author: Luz Paz
Date: 2025-04-09
Found via codespell
#### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
Author: Mike Bradeen
Date: 2025-04-17
Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs,
but the Dial command via ARI did not set an explicit reason. This resulted in a
CANCEL with Normal Call Clearing and corresponding ChannelDestroyed.
This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the
other operations.
Fixes: #963
UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a
CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
#### chan_iax2: Minor improvements to documentation and warning messages.
Author: Naveen Albert
Date: 2025-04-18
* Update Dial() documentation for IAX2 to include syntax for RSA
public key names.
* Add additional details to a couple warnings to provide more context
when an undecodable frame is received.
Resolves: #1206
#### pbx_ael: unregister AELSub application and CLI commands on module load failure
Author: Andreas Wehrmann
Date: 2025-04-18
This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out,
that the AEL module doesn't do proper cleanup when it fails to load.
This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module()
returns an error but load_module() doesn't then unregister CLI cmds and the application.
#### res_pjproject: Fix DTLS client check failing on some platforms
Author: Albrecht Oster
Date: 2025-04-10
Certain platforms (mainly BSD derivatives) have an additional length
field in `sockaddr_in6` and `sockaddr_in`.
`ast_sockaddr_from_pj_sockaddr()` does not take this field into account
when copying over values from the `pj_sockaddr` into the `ast_sockaddr`.
The resulting `ast_sockaddr` will have an uninitialized value for
`sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from
a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()`
has the correct length value set.
This has the effect that `ast_sockaddr_cmp()` will always indicate
an address mismatch, because it does a bitwise comparison, and all DTLS
packets are dropped even if addresses and ports match.
`ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields
are available on the current platform and sets the values accordingly.
Resolves: #505
#### Prequisites for ARI Outbound Websockets
Author: George Joseph
Date: 2025-04-16
stasis:
* Added stasis_app_is_registered().
* Added stasis_app_control_mark_failed().
* Added stasis_app_control_is_failed().
* Fixed res_stasis_device_state so unsubscribe all works properly.
* Modified stasis_app_unregister() to unsubscribe from all event sources.
* Modified stasis_app_exec to return -1 if stasis_app_control_is_failed()
returns true.
http:
* Added ast_http_create_basic_auth_header().
md5:
* Added define for MD5_DIGEST_LENGTH.
tcptls:
* Added flag to ast_tcptls_session_args to suppress connection log messages
to give callers more control over logging.
http_websocket:
* Add flag to ast_websocket_client_options to suppress connection log messages
to give callers more control over logging.
* Added username and password to ast_websocket_client_options to support
outbound basic authentication.
* Added ast_websocket_result_to_str().
#### contrib: Add systemd service and timer files for malloc trim.
Author: Ben Ford
Date: 2025-04-16
Adds two files to the contrib/systemd/ directory that can be installed
to periodically run "malloc trim" on Asterisk. These files do nothing
unless they are explicitly moved to the correct location on the system.
Users who are experiencing Asterisk memory issues can use this service
to potentially help combat the problem. These files can also be
configured to change the start time and interval. See systemd.timer(5)
and systemd.time(7) for more information.
UserNote: Service and timer files for systemd have been added to the
contrib/systemd/ directory. If you are experiencing memory issues,
install these files to have "malloc trim" periodically run on the
system.
#### action_redirect: remove after_bridge_goto_info
Author: Peter Jannesen
Date: 2025-03-13
Under certain circumstances the context/extens/prio are stored in the
after_bridge_goto_info. This info is used when the bridge is broken by
for hangup of the other party. In the situation that the bridge is
broken by an AMI Redirect this info is not used but also not removed.
With the result that when the channel is put back in a bridge and the
bridge is broken the execution continues at the wrong
context/extens/prio.
Resolves: #1144
#### channel: Always provide cause code in ChannelHangupRequest.
Author: Joshua C. Colp
Date: 2025-04-16
When queueing a channel to be hung up a cause code can be
specified in one of two ways:
1. ast_queue_hangup_with_cause
This function takes in a cause code and queues it as part
of the hangup request, which ultimately results in it being
set on the channel.
2. ast_channel_hangupcause_set + ast_queue_hangup
This combination sets the hangup cause on the channel before
queueing the hangup instead of as part of that process.
In the #2 case the ChannelHangupRequest event would not contain
the cause code. For consistency if a cause code has been set
on the channel it will now be added to the event.
Resolves: #1197
#### Add log-caller-id-name option to log Caller ID Name in queue log
Author: phoneben
Date: 2025-02-28
Add log-caller-id-name option to log Caller ID Name in queue log
This patch introduces a new global configuration option, log-caller-id-name,
to queues.conf to control whether the Caller ID name is logged when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is logged
as parameter 4 in the queue log, provided it’s allowed by the
existing log_restricted_caller_id rules. If log-caller-id-name=no (the default),
the Caller ID name is omitted from the logs.
Fixes: #1091
UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
Any '|' characters in the caller ID name will be replaced with '_'.
(provided it’s allowed by the existing log_restricted_caller_id rules).
When log-caller-id-name=no (the default), the Caller ID name is omitted.
#### asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
Author: George Joseph
Date: 2025-04-10
Commands in the "[startup_commands]" section of cli.conf have historically run
after all core and module initialization has been completed and just before
"Asterisk Ready" is printed on the console. This meant that if you
wanted to debug initialization of a specific module, your only option
was to turn on debug for everything by setting "debug" in asterisk.conf.
This commit introduces options to allow you to run CLI commands earlier in
the asterisk startup process.
A command with a value of "pre-init" will run just after logger initialization
but before most core, and all module, initialization.
A command with a value of "pre-module" will run just after all core
initialization but before all module initialization.
A command with a value of "fully-booted" (or "yes" for backwards
compatibility) will run as they always have been...after all
initialization and just before "Asterisk Ready" is printed on the console.
This means you could do this...
```
[startup_commands]
core set debug 3 res_pjsip.so = pre-module
core set debug 0 res_pjsip.so = fully-booted
```
This would turn debugging on for res_pjsip.so to catch any module
initialization debug messages then turn it off again after the module is
loaded.
UserNote: In cli.conf, you can now define startup commands that run before
core initialization and before module initialization.
#### app_confbridge: Prevent crash when publishing channel-less event.
Author: Sean Bright
Date: 2025-04-07
Resolves: #1190
#### ari_websockets: Fix frack if ARI config fails to load.
Author: George Joseph
Date: 2025-04-02
ari_ws_session_registry_dtor() wasn't checking that the container was valid
before running ao2_callback on it to shutdown registered sessions.
#### ARI: REST over Websocket
Author: George Joseph
Date: 2025-03-12
This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
For full details on how to use the new capability, visit...
https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
Changes:
* Added utilities to http.c:
* ast_get_http_method_from_string().
* ast_http_parse_post_form().
* Added utilities to json.c:
* ast_json_nvp_array_to_ast_variables().
* ast_variables_to_json_nvp_array().
* Added definitions for new events to carry REST responses.
* Created res/ari/ari_websocket_requests.c to house the new request handlers.
* Moved non-event specific code out of res/ari/resource_events.c into
res/ari/ari_websockets.c
* Refactored res/res_ari.c to move non-http code out of ast_ari_callback()
(which is http specific) and into ast_ari_invoke() so it can be shared
between both the http and websocket transports.
UpgradeNote: This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
#### audiohook.c: Add ability to adjust volume with float
Author: mkmer
Date: 2025-03-18
Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41.
This is accomplished by the following:
Convert internal variables to type float.
Always use ast_frame_adjust_volume_float() for adjustments.
Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust().
Cast float to int in ast_audiohook_volume_get()
Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float.
This update maintains 100% backward compatibility.
Resolves: #1171
#### audiosocket: added support for DTMF frames
Author: Florent CHAUVEAU
Date: 2025-02-28
Updated the AudioSocket protocol to allow sending DTMF frames.
AST_FRAME_DTMF frames are now forwarded to the server, in addition to
AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF
with value 0x03 was added to the protocol. The payload is a 1-byte
ascii representing the DTMF digit (0-9,*,#...).
UserNote: The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).
#### asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'
Author: Norm Harrison
Date: 2023-04-03
Co-authored-by: Florent CHAUVEAU <florentch@pm.me>
#### audiosocket: fix timeout, fix dialplan app exit, server address in logs
Author: Norm Harrison
Date: 2023-04-03
- Correct wait timeout logic in the dialplan application.
- Include server address in log messages for better traceability.
- Allow dialplan app to exit gracefully on hangup messages and socket closure.
- Optimize I/O by reducing redundant read()/write() operations.
Co-authored-by: Florent CHAUVEAU <florentch@pm.me>
#### chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip s..
Author: Mark Murawski
Date: 2025-03-23
CLI 'pjsip show contact' does not show enough information.
One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact
This feature adds the same details as PJSIPShowContacts to the CLI
Resolves: #643
#### Update config.guess and config.sub
Author: Zhai Liangliang
Date: 2025-03-26
#### chan_pjsip: set correct Endpoint Device State on multiple channels
Author: Alexei Gradinari
Date: 2025-03-25
1. When one channel is placed on hold, the device state is set to ONHOLD
without checking other channels states.
In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN
to calculate aggregate device state of all active channels.
2. The current implementation incorrectly classifies channels in use.
The only channels that has the states: UP, RING and BUSY are considered as "in use".
A channel should be considered "in use" if its state is anything other than
DOWN or RESERVED.
3. Currently, if the number of channels "in use" is greater than device_state_busy_at,
the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate
device state.
The endpoint device state should be BUSY if the number of channels "in use" is greater
than or equal to device_state_busy_at.
Fixes: #1181
#### file.c: missing "custom" sound files should not generate warning logs
Author: Allan Nathanson
Date: 2025-03-18
With `sounds_search_custom_dir = yes` we first look to see if a sound file
is present in the "custom" sound directory before looking in the standard
sound directories. We should not be issuing a WARNING log message if a
sound cannot be found in the "custom" directory.
Resolves: https://github.com/asterisk/asterisk/issues/1170

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@@ -1,4 +1,4 @@
<html><head><title>Readme for asterisk-21.8.0</title></head><body> <html><head><title>Readme for asterisk-21.9.0-rc1</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1> <h1>The Asterisk(R) Open Source PBX</h1>
<pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community. <pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders. Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.</p>
<p>If you are updating from a previous version of Asterisk, make sure you <p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p> read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-21.8.0.html">Change Logs</a></p> <p><a href="ChangeLogs/ChangeLog-21.9.0-rc1.html">Change Logs</a></p>
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3> <h3>NEW INSTALLATIONS</h3>

2
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@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs. read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](ChangeLogs/ChangeLog-21.8.0.html) [Change Logs](ChangeLogs/ChangeLog-21.9.0-rc1.html)
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
### NEW INSTALLATIONS ### NEW INSTALLATIONS