Change Log for Release asterisk-24.0.0-rc1

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Summary:

User Notes:

Upgrade Notes:

Developer Notes:

Commit Authors:

Issue and Commit Detail:

Closed Issues:

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Commit List:

Commit Details:

Prepare master for Asterisk 24
Author: Ben Ford
Date: 2025-08-13

Update version for Asterisk 24
Author: George Joseph
Date: 2026-08-12

res_geolocation: Fix alembic scripts for MySQL.
Author: George Joseph
Date: 2026-09-17

Depending on the backend storage type, MySQL rows could have a maximum
length of 65535 bytes including storage overhead. The number and size of
the fixed-size string columns in the geoloc_profile table cause this to
be exceeded. Unfortunately, this isn't detected until you try to
actually create the table. To address this, the string columns types
have been changed to TEXT, which both MySQL and PostgreSQL handle
without counting to maximum row size. For long, variable-length strings,
TEXT is also more storage efficient.

MySQL also had an issue with the way the Enums were created. These
were also fixed.

res_pjsip: Qualify all resolved contact targets
Author: Mike Bradeen
Date: 2026-07-27

Previously, PJSIP endpoint qualification was offloaded to pjproject which
used a single, internally selected contact based on DNS results.
This could result in unused SRV, A, or AAAA targets, leaving a contact
marked as unreachable without probing the remaining targets.

Now, we resolve the complete set of records before sending and creating
OPTIONS transactions for every resolved target. The contact will be marked
as reachable as soon as any target succeeds and unreachable only after every
target fails.

Fixes: #1927

UserNote: DNS-backed PJSIP contacts are now considered reachable when any resolved
SRV, A, or AAAA target responds to qualification.

func_env: Fix line counting in FILE function for DOS (CR LF) endings.
Author: Naveen Albert
Date: 2026-09-14

The DOS line counting mode was looking for LF CR, when it should have
been looking for CR LF. As a result, line mode never worked properly
for files with DOS (CR LF) line endings, instead erroneously
triggering an error about the offset being negative.

This bug has been present since line mode was introduced in
commit 50d5f13. LF CR is not
a line ending sequence that exists in any line ending format.

Swap the order around so that DOS mode works properly.
Also clarify some of the documentation around FILE operation.

Resolves: #2164

ARI, chan_websocket, res_websocket_client: Handle non-blocking and timeout correctly.
Author: George Joseph
Date: 2026-09-03

This change was prompted by an issue where if the websocket peer to
which ARI or chan_websocket is connected stops consuming TCP packets,
its TCP receive buffer will begin to fill and ultimately cause the
client's TCP send buffer to start filling. When the send buffer is
completely filled, further writes to the websocket will block. This can
cause a cascading lock situation with chan_websocket because it sends
large amounts of data very quickly but can also affect ARI.

Resolves: #2140

UpgradeNote: A new "write_timeout" parameter has been added to
websocket_client.conf that allows setting the maximum amount of time a
write operation can take before returning an error.

UpgradeNote: A new "write_timeout" parameter has been added to
chan_websocket.conf setting the maximum amount of time a write operation
can take before returning an error. Outgoing/client connections can
override this with the new "write_timeout" parameter that was also added
to websocket_client.conf.

DeveloperNote: A new "write_timeout" field has been added to the
ast_websocket_client_options structure that allows setting the maximum
amount of time a write operation can take before returning an error.
The websocket session must be set to nonblocking for this to take effect.

chan_dahdi: Avoid transitioning device state from INUSE to RINGING.
Author: Naveen Albert
Date: 2026-09-11

By default, DAHDI relies on the Asterisk code to provide most device
state, especially for analog lines, in which case it relies entirely
on the core. When an incoming call arrives to a DAHDI line, it
immediately asks the core to update device state for the channel.

This can sometimes lead to unoptimal outcomes; for example, when
ringing an FXS line (station), the device state is momentarily
"INUSE" and then almost immediately transitions to "RINGING". However,
this is incompatible with the state machines of some CPE. For example,
Polycom IP phones' BLFs will ignore any "RINGING" received after an
"INUSE". This makes sense because a line can't ring after being in use
without first being made idle. (The reverse case, RINGING followed by
INUSE, is allowed, since one can answer a ringing phone.)

Thus, to ensure that BLFs accurately display the status of DAHDI
channels, we need to avoid the momentary "INUSE" device state
for ringing channels.

Resolves: #2159

func_channel: add NULL checks on channel tech for dummy channels
Author: Alexandre Fournier
Date: 2026-07-23

A dummy channel allocated with ast_dummy_channel_alloc() never gets a
channel technology, so ast_channel_tech() returns NULL for it.

func_channel_read() dereferences ast_channel_tech(chan)->type for
CHANNEL(channeltype) without checking the tech for NULL, and
func_channel_write_real() dereferences it the same way in its fallback
branch. Both crash with a SIGSEGV on a dummy channel.

This happens when CHANNEL(channeltype) is present in channelvars of
ari.conf and the variable is evaluated on a dummy channel, e.g. when
a voicemail is deposited for a mailbox that has an email address
configured.

Guard both dereferences. A dummy channel has no technology, so
CHANNEL(channeltype) now reads as an empty string and a write falls
through to the "Unknown or unavailable item requested" warning.

This is very similar to the issue fixed by
#1993

Resolves: #2150

AI disclaimer: Claude Opus 4.8 was used to find the cause of the crash
and to find a solution.

Co-Authored-By: Claude Opus 5 (1M context) noreply@anthropic.com

chan_dahdi: Simplify checks for FXO signaling.
Author: Naveen Albert
Date: 2026-09-09

Many parts of chan_dahdi and sig_analog check if a channel is
FXO-signaled (i.e. an FXS channel) by enumerating all 3 types
of FXO-signaled channels. Add macros to simplify these checks
and improve code readability (sig_analog already has one for
FXS-signaled channels).

Resolves: #2152

app_amd: Correct and clarify the AMD documentation.
Author: Jeremy Lainé
Date: 2026-09-10

Fix the miniumWordLength and betweenWordSilence parameter names, which
did not match the arguments AMD() accepts.

Document the NOAUDIODATA cause, explain how values are appended to
AMDCAUSE, and name the parameter each cause reports. MAXWORDLENGTH
appends a single value, so its phantom second field is removed. Also
note that both variables are set to the empty string when detection
cannot be performed, state the millisecond unit once, and describe
maximumWordLength as continuous voice, which is what the check
measures.

Resolves: #2154

apps/app_chanspy: don't delay spyee Hangup when spy channel has no reverse media
Author: Tinet-mucw
Date: 2026-08-26

channel_spy() waited indefinitely on the spy channel, so after the spyee
hung up the spy audiohook leaving RUNNING was not noticed until waitfor
timed out. That kept an autochan reference on the spyee and delayed its
destructor / Hangup.

Poll with a short timeout instead. On timeout, only re-check hook status;
call ast_read() only when the spy channel is readable.

Resolves: #2123

res_prometheus: add toggle config for metrics
Author: whitetreebug
Date: 2026-08-28

In high‑volume call scenarios, Prometheus exports a large number of channels‑related and bridges-related metrics.Not all users require channels_state, channels_duration and bridges_channels.Add conf options to enable or disable channel_details(covering channels_state, channels_duration) and bridge_details(covering bridges_channels).

UserNote: Add new config option 'channels_detail_metrics_enabled' to disable export of 'asterisk_channels_state' and 'asterisk_channels_duration_seconds' metrics,It defaults to 'yes' to preserve existing behavior

Add new config option 'bridges_detail_metrics_enabled' to disable export of 'asterisk_bridges_channels_count' metric, It defaults to 'yes' to preserve existing behavior

Explicitly enable bridges_detail_metrics_enabled for the bridge_to_string unit‑test.

chan_dahdi: Make DAHDI device state cacheable to fix device state sync.
Author: Naveen Albert
Date: 2026-09-03

Commit 8fb5bdc disabled caching of
certain device states that weren't associated with real devices in
order to prevent a resource exhaustion attack from guest/unauthenticated
connections.

In most cases, code was converted to remain cacheable; the few exceptions
had mainly to do with certain paths in channel drivers for guest or
unauthenticated access. However, everything in chan_dahdi was made
uncachable for no apparent reason. This is not harmless, since it
prevents device state synchronization from working (e.g. with
res_pjsip_publish_asterisk) since device state updates that are
marked uncachable result in hints on remote systems always showing
"Unavailable" since uncached device state updates are simply discarded.

Since DAHDI devices very much correspond to real devices (perhaps more
so than in any other channel driver), we can safely cache device state
for DAHDI channels, which also ensures that device state
synchronization now works properly.

Resolves: #2138

app_confbridge: Add announcements option to disable conference announcements
Author: Fran Vicente
Date: 2026-09-02

Every conference unconditionally created an announcer channel (CBAnn) and a
taskprocessor thread to play announcements to the conference as a whole, even on
systems that never play any. That costs a channel, a thread and a bridge member
per conference.

A new bridge profile option, announcements, defaults to yes and preserves the
previous behavior. When set to no, no announcer channel is created and nothing
is played to the conference as a whole: the join and leave sounds, recorded name
intros, the participant count announced to everyone, sound_begin and
sound_leader_has_left are all suppressed, and users are never prompted to record
their name regardless of the announce_join_leave user profile option.

Prompts played only to the participant that caused them are unaffected, since
they are played on that participant's own channel. sound_kicked, sound_muted,
sound_unmuted, the join sound played back by hear_own_join_sound and the
participant count requested from the DTMF menu all still play.

The announcer channel exists for the lifetime of the conference, so the value
that takes effect is the one from the bridge profile of the channel that creates
the conference.

Resolves: #2131

UserNote: A new ConfBridge bridge profile option, announcements, can be set to
no to stop announcements from being played to the conference as a whole. No
announcer channel is then created, which saves a channel and a taskprocessor
thread per conference. Prompts played only to the participant that caused them
are not affected. It defaults to yes, which is the previous behavior.

Bundled pjproject: Do not discard the ambient CFLAGS
Author: Peter Lemenkov
Date: 2026-08-31

Commit f82393d ("Bundled pjproject: Make it
easier to override options in config_site.h") added

  PJPROJECT_CONFIG_OPTS += CFLAGS="$(PJPROJECT_CFLAGS)"

Previously CFLAGS was never passed on the aconfigure command line, so the
bundled pjproject inherited the exported CFLAGS from the environment. Passing
it explicitly overrides that, and PJPROJECT_CFLAGS contains only the options
this file adds (-DPJ_HAS_LINUX_EPOLL=1), so every distribution flag is lost.

Appending to the ambient CFLAGS rather than replacing them keeps the new
PJPROJECT_CFLAGS override working while restoring inheritance.

Resolves: #2132

Signed-off-by: Peter Lemenkov lemenkov@gmail.com

res_resolver_unbound: Include libunbound error reason
Author: Mehrdad Seifzadeh
Date: 2026-09-06

Failures returned by ub_resolve_async() are currently logged without the
reason provided by libunbound, making resolver configuration failures
difficult to diagnose.

Include ub_strerror() in the error message so the underlying libunbound
failure reason is available in the Asterisk log.

Resolves: #2110

app_queue: Update lastpause on realtime pause transition
Author: seifzadeh
Date: 2026-07-30

Realtime queue members used lastpause to determine whether a member
had transitioned from unpaused to paused. However, lastpause also
stores the timestamp of the member's most recent pause and should not
be cleared when the member is unpaused.

Compare the existing paused state with the new realtime value instead,
and update lastpause only when the member transitions from unpaused to
paused.

Resolves: #1760

chan_websocket: Add "unbuffered" mode and fix a few bugs.
Author: George Joseph
Date: 2026-08-25

Besides the User and Developer notes below, checks have been added to
reject a START_MEDIA_BUFFERING buffering command if it's already
active and to reject a STOP_MEDIA_BUFFERING command if a corresponding
START_MEDIA_BUFFERING command was never issued.

Resolves: #2115

UserNote: A new unbuffered chan_websocket mode has been added which
sends frames to the core as they're received from the WebSocket and your
app. It's similar to passthrough mode where your app is responsible
for proper framing and timing except that, assuming you're not using an
unbufferable codec (opus, speex, g729), in unbuffered mode, you can
still use START_MEDIA_BUFFERING to send bulk media and have
chan_websocket frame and time it. When you send STOP_MEDIA_BUFFERING,
the channel will go back into unbuffered mode. Unbuffered mode is most
useful when the media your app normally sends to Asterisk is already
properly framed and timed but you may occasionally need to send bulk
media. The mode is set with the u dialstring option.

DeveloperNote: Two new fields have been added to the chan_websocket
MEDIA_START event for both the plain text and JSON formats:
passthrough: true/false
unbuffered: true/false
These indicate the mode the channel was created with and correspond
to the p and u dialstring options.

DeveloperNote: The MARK_MEDIA chan_websocket command can now be used
by an app when in "passthrough" mode.

res_geolocation: Create alembic scripts and squash a SEGV.
Author: George Joseph
Date: 2026-08-31

If a profile contained a location_reference that pointed to a location that
didn't exist, a segfault would occur when running the geoloc show profiles
command. The command now prints a notice that there was an issue retrieving
the profile as well as an ERROR log message with the explanation.

UpgradeNote: An Alembic script has been added for res_geolocation that creates
the geoloc_location and geoloc_profile tables.

res_rtp_asterisk: Fix reported packet loss to use current report's difference
Author: Mike Bradeen
Date: 2026-08-18

The lost packet calculation was using the cumulative number of lost packets instead
of the current interval's lost packets. This lead to the MES caculation skewing
over time. We now keep track of the last reported value so we can determine the
current value vs the cumulative value.

Fixes: #2101

res_rtp_asterisk: rtt stored before validation
Author: Mike Bradeen
Date: 2026-08-18

The rtcp rtt value was updated before being validated. The calculating function
was correctly returning an error, but only after setting the value. This could
result in AMI and ARI events being generated with an invalid value.

Fixes: #2099

res_rtp_asterisk: Correct effective latency calculation
Author: Mike Bradeen
Date: 2026-08-18

The effective latency calculation was using the full round trip time instead of
the one-way, which causes the MES score to be worse than it would otherwise.

Additionally, the rx jitter calculation was not scaling the normdev and stdev
values to milliseconds, essentially disregarding their values in the calculation

Fixes: #2097

res_rtp_asterisk: Change MES calculation to use long-term jitter mean
Author: Mike Bradeen
Date: 2026-08-18

The MES score used the reported jitter value instead of the smoothed statistic
value, which could cause a single high anomalous RTCP sample to unduly skew
the overall score.

Fixes: #2095

res_websocket_client: Release lock_obj when retries are exhausted
Author: Gian Diego Javes
Date: 2026-08-25

ast_websocket_client_connect() takes lock_obj at the top of every pass
through the retry loop. When the attempts run out it leaves through the
break and returns NULL without releasing it, so the calling thread ends
up holding it. The other two exits, the ast_calloc failure and the
success path, do release it.

The retry path also takes the lock again instead of releasing it before
sleeping. The header describes lock_obj as "an ao2 object to lock while
the connection is being attempted", and the usleep is the wait between
attempts rather than an attempt, so that is where it should be released.

chan_websocket.c passes the per-call instance as lock_obj, allocated
with ao2_alloc and no flags, so it gets the default
AO2_ALLOC_OPT_LOCK_MUTEX. When the object is freed, __ao2_ref() reaches
ast_mutex_destroy() with the mutex still held and frees the memory
anyway. There is no deadlock, Asterisk mutexes are recursive, but every
call that fails to connect logs two ERROR lines.

With each pass balanced, the lock is held exactly once when the loop
breaks, so a single unlock before the return is enough.

Fixes: #2121

frame: Correct ast_frame_adjust_volume* documentation of 'adjustment'
Author: Jacky W J Li
Date: 2026-08-25

The documentation for ast_frame_adjust_volume() and
ast_frame_adjust_volume_float() described the 'adjustment' parameter as a
dB value. In reality the parameter is a linear gain factor applied
directly to the audio samples: a positive value multiplies each sample,
a negative value divides each sample, and 0 leaves the audio unchanged.

The documentation block has also been reformatted to use the standard
' *' Doxygen style (asterisk in the second column) for consistency with
the rest of the file.

To convert a dB change to this factor use adjustment = 10^(dB/20)
(for example, 2 ~= +6 dB and 10 ~= +20 dB).

Resolves: #1167

res_config_odbc: Missing SQLFreeHandle in error paths.
Author: Jaco Kroon
Date: 2026-06-18

Signed-off-by: Jaco Kroon jaco@uls.co.za

res_cdrel_custom: Use extendable ast_str for JSON records.
Author: George Joseph
Date: 2026-08-20

Since JSON records include the field names, the actual record size can be quite
lengthy and since the thread-local backed ast_str used by the DSV writer can't be
extended, we need to use a regular, extendable ast_str.

Resolves: #2106

res_geolocation: Don't crash on invalid sorcery configuration
Author: Sean Bright
Date: 2026-08-21

A crash will occur when you've configured object mappings using the
realtime wizard but neglect to set up your realtime configuration in
extconfig.conf. For example, adding the following to sorcery.conf:

  [res_geolocation]
  location = realtime,geo_location
  profile = realtime,geo_profile

But not defining geo_location or geo_profile in extconfig.conf
will cause Asterisk to crash on startup.

res_geolocation: Load when explicitly configured in sorcery.conf
Author: Sean Bright
Date: 2026-08-20

If you've explicitly configured your configuration objects in
sorcery.conf, it is normal for ast_sorcery_apply_default to return
something other than
AST_SORCERY_APPLY_SUCCESS (e.g. AST_SORCERY_APPLY_DEFAULT_UNNECESSARY).

This is one of the few places (outside of unit tests) that we actual
pay attention to the return value of ast_sorcery_apply_default at all.

res_stasis_recording: Allow recording a channel that is dialing via ARI
Author: fsliwenjie
Date: 2026-08-13

When an ARI application dials a channel with channels.dial, the channel
is placed into an internal, invisible holding bridge (the dial bridge)
while dialing. record_file() rejected recording for any channel that was
in a bridge, so channels.record failed with "Cannot record channel while
in bridge" even though the application never joined the channel to a
bridge of its own.

Narrow the check so it only rejects recording when the channel is in a
bridge that is NOT flagged invisible. The internal dial bridge is flagged
AST_BRIDGE_FLAG_INVISIBLE (it is created by ARI internally, not by the
application), so recording is still permitted there. A "real" bridge
created by the application - including a user-created ARI holding bridge

Resolves: #2044

alembic: Preserve sub-second precision in the queue_log time column.
Author: OMAR.A
Date: 2026-08-07

ast_queue_log() formats realtime queue_log timestamps with microseconds
("%F %T.%6q" in main/logger.c), but the queue_log table declares 'time' as
a generic SQLAlchemy DateTime. That renders on MySQL and MariaDB as bare
DATETIME, which has a fractional-seconds precision of 0, so MariaDB
truncates the fraction and MySQL rounds it. Every event of a single queue
entry can collapse onto one second, or move to the next one, and neither
server warns, not even in strict mode.

Once that happens, queue_log rows can no longer be ordered against each
other, or against the CEL records that cel_odbc and cel_pgsql do stamp
with microseconds.

This adds a revision that widens the queue_log column to DATETIME(6) on
MySQL and MariaDB. PostgreSQL is unaffected and left untouched: DateTime
already renders as TIMESTAMP, which keeps microseconds by default.

Fixes: #2072

UpgradeNote: (schema) On MySQL and MariaDB the queue_log 'time' column is widened
from DATETIME to DATETIME(6) so that the microsecond timestamps Asterisk
already writes are stored rather than discarded. This ALTER rewrites the
table, a datetime precision change is neither INSTANT nor INPLACE, so
queue_log writes will block for its duration on a large table. PostgreSQL
installations are unaffected. Rows already written at whole-second precision will remain as-is and won't be updated.

main/file.c: Log the reason a file could not be opened.
Author: Jeremy Lainé
Date: 2026-08-18

When ast_streamfile() failed to open a file it reported strerror(errno),
but errno was not set by anything on that path. The value came from
whatever syscall ran last, so the message routinely blamed an unrelated
error.

Drop errno from that message and log at the point of failure instead:
filehelper() now reports the fopen() error, which does carry a meaningful
errno, and an allocation failure from get_filestream(). Candidate files
whose format does not match the channel are logged at debug level only,
since filehelper() walks every registered format and skipping the ones
that do not match is normal: a sound installed in several formats would
otherwise warn on every successful playback.

Failures inside the format module's open callback stay unlogged here,
as those callbacks already report their own reason.

Resolves: #1629

chore: update .gitignore for clangd
Author: whitetreebug
Date: 2026-08-18

Ignore clangd generated artifacts: compile_commands.json, .cache

Ignores for .clangd, .clang-format, .clang-tidy tooling files

func_speex: Hold channel lock while updating speex state.
Author: Tinet-mucw
Date: 2026-08-17

speex_write() previously unlocked the channel immediately after
looking up the speex datastore, then continued to create, modify,
or destroy SpeexPreprocessState and related direction data without
the lock. speex_callback() runs from the media path with the channel
already locked, so concurrent Set(DENOISE)/Set(AGC) could free or
mutate that state while preprocess was running and crash inside
speex_preprocess_run().
Keep the channel locked for the full configuration update, unlock
before ast_audiohook_attach()/detach(), and hold the lock across
speex_read() while copying values out of the datastore.

Fixes: #2091

pjsip_wizard.conf.sample: Add IP authentication and TLS examples.
Author: Maksym Tushkov
Date: 2026-08-12

The only ITSP example in the file shows a single trunk that registers and
authenticates. Many providers, including DIDWW which motivated this
change, instead authenticate by IP address, deliver inbound calls from
several regional POPs, and terminate outbound calls through a separate
gateway that does require digest authentication. Several things about
configuring that with the wizard are not obvious from the option
reference:

TLS has a pitfall of its own: the wizard cannot create transport objects,
so the transport has to be defined in pjsip.conf and referenced here by
name, while media_encryption belongs to the endpoint and needs the
endpoint/ prefix.

Add commented-out examples for an IP authenticated trunk pair sharing a
template, and for a TLS trunk with SRTP.

No functional change.

UserNote: pjsip_wizard.conf.sample now includes examples for an ITSP that
authenticates by IP address and delivers calls from several hosts, and
for a TLS trunk with SRTP.

res_http_websocket: Check for client before handling PONG frames.
Author: George Joseph
Date: 2026-08-13

websocket_handled_pong_or_close() now checks that session->client is valid
before trying to check missed_pong_count.

Resolves: #2085

res_pjsip_config_wizard: Fix concurrent Named ACL reloads
Author: Michal Hajek
Date: 2026-06-17

Named ACL change notifications reloaded PJSIP sorcery directly from the
Stasis callback. Under concurrent ACL, core, and config wizard reloads, that
could overlap with config wizard object type observers that compare, prune,
destroy, and replace the per-object-type last_config cache.

Route the ACL-triggered reload through the PJSIP servant path and track when
a Named ACL reload is pending or running so the observer still bypasses the
unchanged-file optimization. Also serialize the observer cache update to
prevent overlapping reload paths from mutating last_config at the same time.

UserNote: Fixes a PJSIP config wizard reload hang during overlapping reloads.