<li><ahref="https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2">GHSA-c7p6-7mvq-8jq2</a>: cli_permissions.conf: deny option does not work for disallowing shell commands</li>
<p>When the callback() API was invoked but no channel passed the test, callback
would return the last channel tested instead of NULL. It now correctly
returns NULL when no channel matches.</p>
<p>Resolves: #1288</p>
<h4>audiohook.c: Improve frame pairing logic to avoid MixMonitor breakage with mix..</h4>
<p>Author: Michal Hajek
Date: 2025-05-21</p>
<p>This patch adjusts the read/write synchronization logic in audiohook_read_frame_both()
to better handle calls where participants use different codecs or sample sizes
(e.g., alaw vs G.722). The previous hard threshold of 2 * samples caused MixMonitor
recordings to break or stutter when frames were not aligned between both directions.</p>
<p>The new logic uses a more tolerant limit (1.5 * samples), which prevents audio tearing
without causing excessive buffer overruns. This fix specifically addresses issues
with MixMonitor when recording directly on a channel in a bridge using mixed codecs.</p>
<p>Reported-by: Michal Hajek <ahref="mailto:michal.hajek@daktela.com">michal.hajek@daktela.com</a></p>
<p>Resolves: #1276
Resolves: #1279</p>
<h4>channelstorage_makeopts.xml: Remove errant XML character.</h4>
<p>UserNote: A new STIR/SHAKEN verification option "ignore_sip_date_header" has
been added that when set to true, will cause the verification process to
not consider a missing or invalid SIP "Date" header to be a failure. This
will make the IAT the sole "truth" for Date in the verification process.
The option can be set in the "verification" and "profile" sections of
stir_shaken.conf.</p>
<p>Also fixed a bug in the port match logic.</p>
<p>Resolves: #1251
Resolves: #1271</p>
<h4>app_record: Add RECORDING_INFO function.</h4>
<p>Author: Naveen Albert
Date: 2024-01-22</p>
<p>Add a function that can be used to retrieve info
about a previous recording, such as its duration.</p>
<p>This is being added as a function to avoid possibly
trampling on dialplan variables, and could be extended
to provide other information in the future.</p>
<p>Resolves: #548</p>
<p>UserNote: The RECORDING_INFO function can now be used
to retrieve the duration of a recording.</p>
<h4>app_sms.c: Fix sending and receiving SMS messages in protocol 2</h4>
<p>Author: Itzanh
Date: 2025-04-06</p>
<p>This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2.</p>
<ul>
<li>Fix MORX message reception (from phone to Asterisk) in SMS protocol 2</li>
<li>Fix MTTX message transmission (from Asterisk to phone) in SMS protocol 2</li>
</ul>
<p>One of the bugs caused messages to have random characters and junk appended at the end up to the character limit. Another bug prevented Asterisk from sending messages from Asterisk to the phone at all. A final bug caused the transmission from Asterisk to the phone to take a long time because app_sms.c did not hang up after correctly sending the message, causing the phone to have to time out and hang up in order to complete the message transmission.</p>
<p>This was tested with a Linksys PAP2T and with a GrandStream HT814, sending and receiving messages with Telefónica DOMO Mensajes phones from Telefónica Spain. I had to play with both the network jitter buffer and the dB gain to get it to work. One of my phones required the gain to be set to +3dB for it to work, while another required it to be set to +6dB.</p>
<p>Only MORX and MTTX were tested, I did not test sending and receiving messages to a TelCo SMSC.</p>
<h4>app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal..</h4>
<p>Author: phoneben
Date: 2025-05-26</p>
<p>This update adds support for a new QUEUE_RAISE_PENALTY format: rN</p>
<p>When QUEUE_RAISE_PENALTY is set to rN (e.g., r4), only members whose current penalty
is greater than or equal to the defined min_penalty and less than or equal to max_penalty
will have their penalty raised to N.</p>
<p>Members with penalties outside the min/max range remain unchanged.</p>
<p>Example behaviors:</p>
<p>QUEUE_RAISE_PENALTY=4 → Raise all members with penalty < 4 (existing behavior)
QUEUE_RAISE_PENALTY=r4 → Raise only members with penalty in [min_penalty, max_penalty] to 4</p>
<p>Implementation details:</p>
<p>Adds parsing logic to detect the r prefix and sets the raise_respect_min flag</p>
<p>Modifies the raise logic to skip members outside the defined penalty range when the flag is active</p>
<p>UserNote: This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises
only for members whose current penalty is within the [min_penalty, max_penalty] range.
Members with lower or higher penalties are unaffected.
This behavior is backward-compatible with existing queue rule configurations.</p>
<h4>res_websocket_client: Add more info to the XML documentation.</h4>
<p>Author: George Joseph
Date: 2025-06-05</p>
<p>Added "see-also" links to chan_websocket and ARI Outbound WebSocket and
added an example configuration for each.</p>
<h4>res_odbc: cache_size option to limit the cached connections.</h4>
<p>UserNote: When using res_odbc it should be noted that back-end
connections to the underlying database can now be configured to re-use
the cached connections in a round-robin manner rather than repeatedly
re-using the same connection. This helps to keep connections alive, and
to purge dead connections from the system, thus more dynamically
adjusting to actual load. The downside is that one could keep too many
connections active for a longer time resulting in resource also begin
consumed on the database side.</p>
<h4>res_pjsip: Fix empty <code>ActiveChannels</code> property in AMI responses.</h4>
<p>Author: Sean Bright
Date: 2025-05-27</p>
<p>The logic appears to have been reversed since it was introduced in
05cbf8df.</p>
<p>Resolves: #1254</p>
<h4>ARI Outbound Websockets</h4>
<p>Author: George Joseph
Date: 2025-03-28</p>
<p>Asterisk can now establish websocket sessions <em>to</em> your ARI applications
as well as accepting websocket sessions <em>from</em> them.
Full details: http://s.asterisk.net/ari-outbound-ws</p>
<p>Code change summary:
* Added an ast_vector_string_join() function,
* Added ApplicationRegistered and ApplicationUnregistered ARI events.
* Converted res/ari/config.c to use sorcery to process ari.conf.
* Added the "outbound-websocket" ARI config object.
* Refactored res/ari/ari_websockets.c to handle outbound websockets.
* Refactored res/ari/cli.c for the sorcery changeover.
* Updated res/res_stasis.c for the sorcery changeover.
* Updated apps/app_stasis.c to allow initiating per-call outbound websockets.
* Added CLI commands to manage ARI websockets.
* Added the new "outbound-websocket" object to ari.conf.sample.
* Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml</p>
<p>UserNote: Asterisk can now establish websocket sessions <em>to</em> your ARI applications
as well as accepting websocket sessions <em>from</em> them.
Full details: http://s.asterisk.net/ari-outbound-ws</p>
<h4>res_websocket_client: Create common utilities for websocket clients.</h4>
<p>Author: George Joseph
Date: 2025-05-02</p>
<p>Since multiple Asterisk capabilities now need to create websocket clients
it makes sense to create a common set of utilities rather than making
each of those capabilities implement their own.</p>
<ul>
<li>A new configuration file "websocket_client.conf" is used to store common
client parameters in named configuration sections.</li>
<li>APIs are provided to list and retrieve ast_websocket_client objects created
from the named configurations.</li>
<li>An API is provided that accepts an ast_websocket_client object, connects
to the remote server with retries and returns an ast_websocket object. TLS is
supported as is basic authentication.</li>
<li>An observer can be registered to receive notification of loaded or reloaded
client objects.</li>
<li>An API is provided to compare an existing client object to one just
reloaded and return the fields that were changed. The caller can then decide
what action to take based on which fields changed.</li>
</ul>
<p>Also as part of thie commit, several sorcery convenience macros were created
to make registering common object fields easier.</p>
<p>UserNote: A new module "res_websocket_client" and config file
"websocket_client.conf" have been added to support several upcoming new
capabilities that need common websocket client configuration.</p>
<h4>asterisk.c: Add option to restrict shell access from remote consoles.</h4>
<p>Author: George Joseph
Date: 2025-05-19</p>
<p>UserNote: A new asterisk.conf option 'disable_remote_console_shell' has
been added that, when set, will prevent remote consoles from executing
shell commands using the '!' prefix.</p>
<p>Resolves: #GHSA-c7p6-7mvq-8jq2</p>
<h4>frame.c: validate frame data length is less than samples when adjusting volume</h4>
<p>Author: mkmer
Date: 2025-05-12</p>
<p>Resolves: #1230</p>
<h4>res_audiosocket.c: Add retry mechanism for reading data from AudioSocket</h4>
<p>Author: Sven Kube
Date: 2025-05-13</p>
<p>The added retry mechanism addresses an issue that arises when fragmented TCP
packets are received, each containing only a portion of an AudioSocket packet.
This situation can occur if the external service sending the AudioSocket data
has Nagle's algorithm enabled.</p>
<h4>res_audiosocket.c: Set the TCP_NODELAY socket option</h4>
<p>Author: Sven Kube
Date: 2025-05-13</p>
<p>Disable Nagle's algorithm by setting the TCP_NODELAY socket option.
This reduces latency by preventing delays caused by packet buffering.</p>
<h4>menuselect: Fix GTK menu callbacks for Fedora 42 compatibility</h4>
<p>Author: Thomas B. Clark
Date: 2025-05-12</p>
<p>This patch resolves a build failure in <code>menuselect_gtk.c</code> when running
<code>make menuconfig</code> on Fedora 42. The new version of GTK introduced stricter
type checking for callback signatures.</p>
<p>Changes include:
- Add wrapper functions to match the expected <code>void (*)(void)</code> signature.
- Update <code>menu_items</code> array to use these wrappers.</p>
<p>Fixes: #1243</p>
<h4>jansson: Upgrade version to jansson 2.14.1</h4>
<p>Author: Stanislav Abramenkov
Date: 2025-03-24</p>
<p>UpgradeNote: jansson has been upgraded to 2.14.1. For more
information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.14.1</p>
<p>Resolves: #1178</p>
<h4>pjproject: Increase maximum SDP formats and attribute limits</h4>
<p>Author: Joe Searle
Date: 2025-05-15</p>
<p>Since Chrome 136, using Windows, when initiating a video call the INVITE SDP exceeds the maximum number of allowed attributes, resulting in the INVITE being rejected. This increases the attribute limit and the number of formats allowed when using bundled pjproject.</p>
<p>Fixes: #1240</p>
<h4>manager.c: Invalid ref-counting when purging events</h4>
<p>Author: Nathan Monfils
Date: 2025-05-05</p>
<p>We have a use-case where we generate a <em>lot</em> of events on the AMI, and
then when doing <code>manager show eventq</code> we would see some events which
would linger for hours or days in there. Obviously something was leaking.
Testing allowed us to track down this logic bug in the ref-counting on
the event purge.</p>
<p>Reproducing the bug was not super trivial, we managed to do it in a
production-like load testing environment with multiple AMI consumers.</p>
<p>The race condition itself:</p>
<ol>
<li>something allocates and links <code>session</code></li>
<li><code>purge_sessions</code> iterates over that <code>session</code> (takes ref)</li>
<li><code>purge_session</code> correctly de-referencess that session</li>
<li><code>purge_session</code> re-evaluates the while() loop, taking a reference</li>
<li><code>purge_session</code> exits (<code>n_max > 0</code> is false)</li>
<li>whatever allocated the <code>session</code> deallocates it, but a reference is
now lost since we exited the <code>while</code> loop before de-referencing.</li>
<li>since the destructor is never called, the session->last_ev->usecount
is never decremented, leading to events lingering in the queue</li>
</ol>
<p>The impact of this bug does not seem major. The events are small and do
not seem, from our testing, to be causing meaningful additional CPU
usage. Mainly we wanted to fix this issue because we are internally
adding prometheus metrics to the eventq and those leaked events were
causing the metrics to show garbage data.</p>
<h4>res_pjsip_nat.c: Do not overwrite transfer host</h4>
<p>Author: Mike Bradeen
Date: 2025-05-08</p>
<p>When a call is transfered via dialplan behind a NAT, the
host portion of the Contact header in the 302 will no longer
be over-written with the external NAT IP and will retain the
hostname.</p>
<p>Fixes: #1141</p>
<h4>chan_pjsip: Serialize INVITE creation on DTMF attended transfer</h4>
<p>Author: Mike Bradeen
Date: 2025-05-05</p>
<p>When a call is transfered via DTMF feature code, the Transfer Target and
Transferer are bridged immediately. This opens the possibilty of a race
condition between the creation of an INVITE and the bridge induced colp
update that can result in the set caller ID being over-written with the
