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<html><head><title>ChangeLog for asterisk-21.12.0-rc1</title></head><body>
<h2>Change Log for Release asterisk-21.12.0-rc1</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0-rc1.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0-rc1">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0-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: 19</li>
<li>Commit Authors: 10</li>
<li>Issues Resolved: 12</li>
<li>Security Advisories Resolved: 0</li>
</ul>
<h3>User Notes:</h3>
<ul>
<li>
<h4>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</h4>
<p>Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.</p>
</li>
<li>
<h4>chan_dahdi: Add DAHDI_CHANNEL function.</h4>
<p>The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.</p>
</li>
</ul>
<h3>Upgrade Notes:</h3>
<ul>
<li>
<h4>res_audiosocket: add message types for all slin sample rates</h4>
New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.</li>
</ul>
<h3>Developer Notes:</h3>
<h3>Commit Authors:</h3>
<ul>
<li>Bastian Triller: (1)</li>
<li>Ben Ford: (1)</li>
<li>George Joseph: (3)</li>
<li>Igor Goncharovsky: (1)</li>
<li>Max Grobecker: (1)</li>
<li>Nathan Monfils: (1)</li>
<li>Naveen Albert: (4)</li>
<li>Phoneben: (1)</li>
<li>Sean Bright: (3)</li>
<li>Sven Kube: (3)</li>
</ul>
<h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3>
<ul>
<li>1340: [bug]: comfort noise packet corrupted</li>
<li>1419: [bug]: static code analysis issues in app_adsiprog.c</li>
<li>1422: [bug]: static code analysis issues in apps/app_externalivr.c</li>
<li>1425: [bug]: static code analysis issues in apps/app_queue.c</li>
<li>1434: [improvement]: pbx_variables: Create real channel for dialplan eval CLI command</li>
<li>1436: [improvement]: res_cliexec: Avoid unnecessary cast to char*</li>
<li>1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function</li>
<li>1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with PJSIP_TRANSFER_HANDLING(ari-only)</li>
<li>1491: [bug]: Segfault: <code>channelstorage_cpp</code> fast lookup without lock (<code>get_by_name_exact</code>/<code>get_by_uniqueid</code>) leads to UAF during hangup</li>
<li>1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message</li>
<li>1539: [bug]: safe_asterisk without TTY doesn't log to file</li>
<li>1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361</li>
</ul>
<h3>Commits By Author:</h3>
<ul>
<li>
<h4>Bastian Triller (1):</h4>
</li>
<li>
<p>Fix some doxygen, typos and whitespace</p>
</li>
<li>
<h4>Ben Ford (1):</h4>
</li>
<li>
<p>rtp_engine.c: Add exception for comfort noise payload.</p>
</li>
<li>
<h4>George Joseph (3):</h4>
</li>
<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</li>
<li>chan_websocket.c: Change payload references to command instead.</li>
<li>
<p>safe_asterisk: Fix logging and sorting issue.</p>
</li>
<li>
<h4>Igor Goncharovsky (1):</h4>
</li>
<li>
<p>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</p>
</li>
<li>
<h4>Max Grobecker (1):</h4>
</li>
<li>
<p>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</p>
</li>
<li>
<h4>Nathan Monfils (1):</h4>
</li>
<li>
<p>manager.c: Fix presencestate object leak</p>
</li>
<li>
<h4>Naveen Albert (4):</h4>
</li>
<li>pbx_variables.c: Create real channel for "dialplan eval function".</li>
<li>res_cliexec: Remove unnecessary casts to char*.</li>
<li>app_adsiprog: Fix possible NULL dereference.</li>
<li>
<p>chan_dahdi: Add DAHDI_CHANNEL function.</p>
</li>
<li>
<h4>Sean Bright (3):</h4>
</li>
<li>audiohook.c: Ensure correct AO2 reference is dereffed.</li>
<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li>
<p>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</p>
</li>
<li>
<h4>Sven Kube (3):</h4>
</li>
<li>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..</li>
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
<li>
<p>res_audiosocket: add message types for all slin sample rates</p>
</li>
<li>
<h4>phoneben (1):</h4>
</li>
<li>app_queue: Add NULL pointer checks in app_queue</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</li>
<li>safe_asterisk: Fix logging and sorting issue.</li>
<li>res_audiosocket: add message types for all slin sample rates</li>
<li>chan_websocket.c: Change payload references to command instead.</li>
<li>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</li>
<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</li>
<li>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</li>
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
<li>Fix some doxygen, typos and whitespace</li>
<li>app_queue: Add NULL pointer checks in app_queue</li>
<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
<li>app_adsiprog: Fix possible NULL dereference.</li>
<li>manager.c: Fix presencestate object leak</li>
<li>audiohook.c: Ensure correct AO2 reference is dereffed.</li>
<li>res_cliexec: Remove unnecessary casts to char*.</li>
<li>rtp_engine.c: Add exception for comfort noise payload.</li>
<li>pbx_variables.c: Create real channel for "dialplan eval function".</li>
</ul>
<h3>Commit Details:</h3>
<h4>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</h4>
<p>Author: Sean Bright
Date: 2025-10-22</p>
<ul>
<li>Using <code>==</code> with the POSIX sh <code>test</code> utility is UB.</li>
<li>Switch back to using globs instead of using <code>$(find … | sort)</code>.</li>
<li>Fix a missing redirect when checking for the OS type.</li>
</ul>
<p>Resolves: #1554</p>
<h4>safe_asterisk: Fix logging and sorting issue.</h4>
<p>Author: George Joseph
Date: 2025-10-17</p>
<p>Re-enabled "TTY=9" which was erroneously disabled as part of a recent
security fix and removed another logging "fix" that was added.</p>
<p>Also added a sort to the "find" that enumerates the scripts to be sourced so
they're sourced in the correct order.</p>
<p>Resolves: #1539</p>
<h4>res_audiosocket: add message types for all slin sample rates</h4>
<p>Author: Sven Kube
Date: 2025-10-10</p>
<p>Extend audiosocket messages with types 0x11 - 0x18 to create asterisk
frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 format, enabling the transmission of audio at a higher sample
rates. For audiosocket messages sent by Asterisk, the message kind is
determined by the format of the originating asterisk frame.</p>
<p>UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.</p>
<h4>chan_websocket.c: Change payload references to command instead.</h4>
<p>Author: George Joseph
Date: 2025-10-08</p>
<p>Some of the tests in process_text_message() were still comparing to the
websocket message payload instead of the "command" string.</p>
<p>Resolves: #1525</p>
<h4>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</h4>
<p>Author: Igor Goncharovsky
Date: 2025-09-04</p>
<p>As soon as SIP call may end with several Reason headers, we
want to make all of them available through the HAGUPCAUSE() function.
This implementation uses the same ao2 hash for cause codes storage
and adds a flag to make difference between last processed sip
message and content of reason headers.</p>
<p>UserNote: Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.</p>
<h4>channelstorage_cpp_map_name_id: Add read locking around retrievals.</h4>
<p>Author: George Joseph
Date: 2025-10-01</p>
<p>When we retrieve a channel from a C++ map, we actually get back a wrapper
object that points to the channel then right after we retrieve it, we bump its
reference count. There's a tiny chance however that between those two
statements a delete and/or unref might happen which would cause the wrapper
object or the channel itself to become invalid resulting in a SEGV. To avoid
this we now perform a read lock on the driver around those statements.</p>
<p>Resolves: #1491</p>
<h4>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</h4>
<p>Author: Max Grobecker
Date: 2025-09-21</p>
<p>This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation.
The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header,
but that option had no effect as it was not implemented by res_pjsip_geolocation.</p>
<p>If the <code>location_source</code> configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).</p>
<p>This commits adds already documented functionality.</p>
<h4>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</h4>
<p>Author: Sven Kube
Date: 2025-09-22</p>
<p>When handling SIP transfers via ARI, there is no protocol_id in case of
a blind transfer.</p>
<p>Resolves: #1467</p>
<h4>Fix some doxygen, typos and whitespace</h4>
<p>Author: Bastian Triller
Date: 2025-09-21</p>
<h4>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..</h4>
<p>Author: Sven Kube
Date: 2025-09-18</p>
<p>When handling SIP transfers via ARI, the <code>referred_by</code> field in
<code>transfer_ari_state</code> may be null, since SIP REFER requests are not
required to include a <code>Referred-By</code> header. Without this check, a null
value caused the transfer to fail and triggered a NOTIFY with a 500
Internal Server Error.</p>
<h4>app_queue: Add NULL pointer checks in app_queue</h4>
<p>Author: phoneben
Date: 2025-09-11</p>
<p>Add NULL check for word_list before calling word_in_list()
Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()</p>
<p>Resolves: #1425</p>
<h4>app_externalivr: Prevent out-of-bounds read during argument processing.</h4>
<p>Author: Sean Bright
Date: 2025-09-17</p>
<p>Resolves: #1422</p>
<h4>chan_dahdi: Add DAHDI_CHANNEL function.</h4>
<p>Author: Naveen Albert
Date: 2025-09-11</p>
<p>Add a dialplan function that can be used to get/set properties of
DAHDI channels (as opposed to Asterisk channels). This exposes
properties that were not previously available, allowing for certain
operations to now be performed in the dialplan.</p>
<p>Resolves: #1455</p>
<p>UserNote: The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.</p>
<h4>app_adsiprog: Fix possible NULL dereference.</h4>
<p>Author: Naveen Albert
Date: 2025-09-10</p>
<p>get_token can return NULL, but process_token uses this result without
checking for NULL; as elsewhere, check for a NULL result to avoid
possible NULL dereference.</p>
<p>Resolves: #1419</p>
<h4>manager.c: Fix presencestate object leak</h4>
<p>Author: Nathan Monfils
Date: 2025-09-08</p>
<p>ast_presence_state allocates subtype and message. We straightforwardly
need to clean those up.</p>
<h4>audiohook.c: Ensure correct AO2 reference is dereffed.</h4>
<p>Author: Sean Bright
Date: 2025-09-10</p>
<p>Part of #1440.</p>
<h4>res_cliexec: Remove unnecessary casts to char*.</h4>
<p>Author: Naveen Albert
Date: 2025-09-09</p>
<p>Resolves: #1436</p>
<h4>rtp_engine.c: Add exception for comfort noise payload.</h4>
<p>Author: Ben Ford
Date: 2025-09-09</p>
<p>In a previous commit, a change was made to
ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample
rates. This ended up returning an invalid payload int for comfort noise.
A check has been added that returns early if the payload is in fact
supposed to be comfort noise.</p>
<p>Fixes: #1340</p>
<h4>pbx_variables.c: Create real channel for "dialplan eval function".</h4>
<p>Author: Naveen Albert
Date: 2025-09-09</p>
<p>"dialplan eval function" has been using a dummy channel for function
evaluation, much like many of the unit tests. However, sometimes, this
can cause issues for functions that are not expecting dummy channels.
As an example, ast_channel_tech(chan) is NULL on such channels, and
ast_channel_tech(chan)-&gt;type consequently results in a NULL dereference.
Normally, functions do not worry about this since channels executing
dialplan aren't dummy channels.</p>
<p>While some functions are better about checking for these sorts of edge
cases, use a real channel with a dummy technology to make this CLI
command inherently safe for any dialplan function that could be evaluated
from the CLI.</p>
<p>Resolves: #1434</p>
</body></html>

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## Change Log for Release asterisk-21.12.0-rc1
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0-rc1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0-rc1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0-rc1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary:
- Commits: 19
- Commit Authors: 10
- Issues Resolved: 12
- Security Advisories Resolved: 0
### User Notes:
- #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.
- #### chan_dahdi: Add DAHDI_CHANNEL function.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
### Upgrade Notes:
- #### res_audiosocket: add message types for all slin sample rates
New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.
### Developer Notes:
### Commit Authors:
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (3)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Phoneben: (1)
- Sean Bright: (3)
- Sven Kube: (3)
## Issue and Commit Detail:
### Closed Issues:
- 1340: [bug]: comfort noise packet corrupted
- 1419: [bug]: static code analysis issues in app_adsiprog.c
- 1422: [bug]: static code analysis issues in apps/app_externalivr.c
- 1425: [bug]: static code analysis issues in apps/app_queue.c
- 1434: [improvement]: pbx_variables: Create real channel for dialplan eval CLI command
- 1436: [improvement]: res_cliexec: Avoid unnecessary cast to char*
- 1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function
- 1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with PJSIP_TRANSFER_HANDLING(ari-only)
- 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock (`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup
- 1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message
- 1539: [bug]: safe_asterisk without TTY doesn't log to file
- 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361
### Commits By Author:
- #### Bastian Triller (1):
- Fix some doxygen, typos and whitespace
- #### Ben Ford (1):
- rtp_engine.c: Add exception for comfort noise payload.
- #### George Joseph (3):
- channelstorage_cpp_map_name_id: Add read locking around retrievals.
- chan_websocket.c: Change payload references to command instead.
- safe_asterisk: Fix logging and sorting issue.
- #### Igor Goncharovsky (1):
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
- #### Max Grobecker (1):
- res_pjsip_geolocation: Add support for Geolocation loc-src parameter
- #### Nathan Monfils (1):
- manager.c: Fix presencestate object leak
- #### Naveen Albert (4):
- pbx_variables.c: Create real channel for "dialplan eval function".
- res_cliexec: Remove unnecessary casts to char*.
- app_adsiprog: Fix possible NULL dereference.
- chan_dahdi: Add DAHDI_CHANNEL function.
- #### Sean Bright (3):
- audiohook.c: Ensure correct AO2 reference is dereffed.
- app_externalivr: Prevent out-of-bounds read during argument processing.
- safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.
- #### Sven Kube (3):
- stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..
- stasis_channels.c: Make protocol_id optional to enable blind transfer via ari
- res_audiosocket: add message types for all slin sample rates
- #### phoneben (1):
- app_queue: Add NULL pointer checks in app_queue
### Commit List:
- safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.
- safe_asterisk: Fix logging and sorting issue.
- res_audiosocket: add message types for all slin sample rates
- chan_websocket.c: Change payload references to command instead.
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
- channelstorage_cpp_map_name_id: Add read locking around retrievals.
- res_pjsip_geolocation: Add support for Geolocation loc-src parameter
- stasis_channels.c: Make protocol_id optional to enable blind transfer via ari
- Fix some doxygen, typos and whitespace
- app_queue: Add NULL pointer checks in app_queue
- app_externalivr: Prevent out-of-bounds read during argument processing.
- chan_dahdi: Add DAHDI_CHANNEL function.
- app_adsiprog: Fix possible NULL dereference.
- manager.c: Fix presencestate object leak
- audiohook.c: Ensure correct AO2 reference is dereffed.
- res_cliexec: Remove unnecessary casts to char*.
- rtp_engine.c: Add exception for comfort noise payload.
- pbx_variables.c: Create real channel for "dialplan eval function".
### Commit Details:
#### safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.
Author: Sean Bright
Date: 2025-10-22
* Using `==` with the POSIX sh `test` utility is UB.
* Switch back to using globs instead of using `$(find … | sort)`.
* Fix a missing redirect when checking for the OS type.
Resolves: #1554
#### safe_asterisk: Fix logging and sorting issue.
Author: George Joseph
Date: 2025-10-17
Re-enabled "TTY=9" which was erroneously disabled as part of a recent
security fix and removed another logging "fix" that was added.
Also added a sort to the "find" that enumerates the scripts to be sourced so
they're sourced in the correct order.
Resolves: #1539
#### res_audiosocket: add message types for all slin sample rates
Author: Sven Kube
Date: 2025-10-10
Extend audiosocket messages with types 0x11 - 0x18 to create asterisk
frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 format, enabling the transmission of audio at a higher sample
rates. For audiosocket messages sent by Asterisk, the message kind is
determined by the format of the originating asterisk frame.
UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.
#### chan_websocket.c: Change payload references to command instead.
Author: George Joseph
Date: 2025-10-08
Some of the tests in process_text_message() were still comparing to the
websocket message payload instead of the "command" string.
Resolves: #1525
#### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
Author: Igor Goncharovsky
Date: 2025-09-04
As soon as SIP call may end with several Reason headers, we
want to make all of them available through the HAGUPCAUSE() function.
This implementation uses the same ao2 hash for cause codes storage
and adds a flag to make difference between last processed sip
message and content of reason headers.
UserNote: Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.
#### channelstorage_cpp_map_name_id: Add read locking around retrievals.
Author: George Joseph
Date: 2025-10-01
When we retrieve a channel from a C++ map, we actually get back a wrapper
object that points to the channel then right after we retrieve it, we bump its
reference count. There's a tiny chance however that between those two
statements a delete and/or unref might happen which would cause the wrapper
object or the channel itself to become invalid resulting in a SEGV. To avoid
this we now perform a read lock on the driver around those statements.
Resolves: #1491
#### res_pjsip_geolocation: Add support for Geolocation loc-src parameter
Author: Max Grobecker
Date: 2025-09-21
This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation.
The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header,
but that option had no effect as it was not implemented by res_pjsip_geolocation.
If the `location_source` configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).
This commits adds already documented functionality.
#### stasis_channels.c: Make protocol_id optional to enable blind transfer via ari
Author: Sven Kube
Date: 2025-09-22
When handling SIP transfers via ARI, there is no protocol_id in case of
a blind transfer.
Resolves: #1467
#### Fix some doxygen, typos and whitespace
Author: Bastian Triller
Date: 2025-09-21
#### stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..
Author: Sven Kube
Date: 2025-09-18
When handling SIP transfers via ARI, the `referred_by` field in
`transfer_ari_state` may be null, since SIP REFER requests are not
required to include a `Referred-By` header. Without this check, a null
value caused the transfer to fail and triggered a NOTIFY with a 500
Internal Server Error.
#### app_queue: Add NULL pointer checks in app_queue
Author: phoneben
Date: 2025-09-11
Add NULL check for word_list before calling word_in_list()
Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()
Resolves: #1425
#### app_externalivr: Prevent out-of-bounds read during argument processing.
Author: Sean Bright
Date: 2025-09-17
Resolves: #1422
#### chan_dahdi: Add DAHDI_CHANNEL function.
Author: Naveen Albert
Date: 2025-09-11
Add a dialplan function that can be used to get/set properties of
DAHDI channels (as opposed to Asterisk channels). This exposes
properties that were not previously available, allowing for certain
operations to now be performed in the dialplan.
Resolves: #1455
UserNote: The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
#### app_adsiprog: Fix possible NULL dereference.
Author: Naveen Albert
Date: 2025-09-10
get_token can return NULL, but process_token uses this result without
checking for NULL; as elsewhere, check for a NULL result to avoid
possible NULL dereference.
Resolves: #1419
#### manager.c: Fix presencestate object leak
Author: Nathan Monfils
Date: 2025-09-08
ast_presence_state allocates subtype and message. We straightforwardly
need to clean those up.
#### audiohook.c: Ensure correct AO2 reference is dereffed.
Author: Sean Bright
Date: 2025-09-10
Part of #1440.
#### res_cliexec: Remove unnecessary casts to char*.
Author: Naveen Albert
Date: 2025-09-09
Resolves: #1436
#### rtp_engine.c: Add exception for comfort noise payload.
Author: Ben Ford
Date: 2025-09-09
In a previous commit, a change was made to
ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample
rates. This ended up returning an invalid payload int for comfort noise.
A check has been added that returns early if the payload is in fact
supposed to be comfort noise.
Fixes: #1340
#### pbx_variables.c: Create real channel for "dialplan eval function".
Author: Naveen Albert
Date: 2025-09-09
"dialplan eval function" has been using a dummy channel for function
evaluation, much like many of the unit tests. However, sometimes, this
can cause issues for functions that are not expecting dummy channels.
As an example, ast_channel_tech(chan) is NULL on such channels, and
ast_channel_tech(chan)->type consequently results in a NULL dereference.
Normally, functions do not worry about this since channels executing
dialplan aren't dummy channels.
While some functions are better about checking for these sorts of edge
cases, use a real channel with a dummy technology to make this CLI
command inherently safe for any dialplan function that could be evaluated
from the CLI.
Resolves: #1434

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@@ -1,4 +1,4 @@
<html><head><title>Readme for asterisk-21.11.0</title></head><body>
<html><head><title>Readme for asterisk-21.12.0-rc1</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1>
<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.
@@ -37,7 +37,7 @@ hardware.</p>
<p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-21.11.0.html">Change Logs</a></p>
<p><a href="ChangeLogs/ChangeLog-21.12.0-rc1.html">Change Logs</a></p>
<!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3>

2
README.md
View File

@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](ChangeLogs/ChangeLog-21.11.0.html)
[Change Logs](ChangeLogs/ChangeLog-21.12.0-rc1.html)
<!-- END-CHANGELOGS -->
### NEW INSTALLATIONS