From a8d92e089d0692f729a273b991c2cdefbb571e5d Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Thu, 4 Jun 2026 14:02:38 +0000 Subject: [PATCH] Update for 22.10.0-rc1 --- .version | 2 +- CHANGES.html | 2 +- CHANGES.md | 2 +- ChangeLogs/ChangeLog-22.10.0-rc1.html | 1202 +++++++++++++++ ChangeLogs/ChangeLog-22.10.0-rc1.md | 1336 +++++++++++++++++ README.html | 4 +- README.md | 2 +- contrib/realtime/mysql/mysql_config.sql | 8 + .../realtime/postgresql/postgresql_config.sql | 8 + 9 files changed, 2560 insertions(+), 6 deletions(-) create mode 100644 ChangeLogs/ChangeLog-22.10.0-rc1.html create mode 100644 ChangeLogs/ChangeLog-22.10.0-rc1.md diff --git a/.version b/.version index 728f7de5c2..1c27fd0c46 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -22.9.0 +22.10.0-rc1 diff --git a/CHANGES.html b/CHANGES.html index 600b5e5a8d..9ce1511912 120000 --- a/CHANGES.html +++ b/CHANGES.html @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.9.0.html \ No newline at end of file +ChangeLogs/ChangeLog-22.10.0-rc1.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index b2ebcf104e..1de145bee1 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.9.0.md \ No newline at end of file +ChangeLogs/ChangeLog-22.10.0-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-22.10.0-rc1.html b/ChangeLogs/ChangeLog-22.10.0-rc1.html new file mode 100644 index 0000000000..7eb1d775ea --- /dev/null +++ b/ChangeLogs/ChangeLog-22.10.0-rc1.html @@ -0,0 +1,1202 @@ +ChangeLog for asterisk-22.10.0-rc1 +

Change Log for Release asterisk-22.10.0-rc1

+

Links:

+ +

Summary:

+ +

User Notes:

+ +

Upgrade Notes:

+ +

Developer Notes:

+ +

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

Commit Details:

+

res_ari: Add res_ari_model as an optional_module.

+

Author: George Joseph + Date: 2026-06-03

+

Under certain timing/load conditions, res_ari_model may not load until after + res_ari on startup or it might unload before res_ari on shutdown. This can + cause a segfault when DEVMODE is enabled and there are persistent outbound + websocket connections because DEVMODE forces validation of outgoing events + against the models. To prevent this, res_ari_model has been added as an + "optional_module" to res_ari's NODULE_INFO. This will enforce load/unload + order but not make res_ari dependent on res_ari_model. However, if + Asterisk is configured with --enable-dev-mode, res_ari will fail to + load if res_ari_model isn't available.

+

Resolves: #1970

+

res ari: Add attachable states to Channels and Bridges

+

Author: Mike Bradeen + Date: 2026-03-31

+

Adds the ability to attach multiple states to both Channels and Bridges in the form + of variables that are included in all events on the associated object.

+

First, this adds an optional boolean field to channel variables 'report_events' + that causes the variable to automatically be included in all events on that channel.

+

To allow this, variables can now be either name value pairs (the current format): + <variable_name>: '<value_string>' + - or - + <variable_name>: {value: '<value_string>', report_events: [true|false]}

+

If the old format is used or 'report_events' is not included, it will default to + false and retain current behavior.

+

Second, this extends both reported and unreported variables to Bridges so they too + may have stateful information.

+

Resolves: #1910

+

UserNote: Bridge variables now can be set and retrieved via the following paths: + /bridges/{bridgeId}/variable + /bridges/{bridgeId}/variables + Both Bridge and Channel variables can now be set with an optional 'report_events' + boolean flag that will cause those variables to be included on all events on that + object. The 'report_events' flag will default to False if not set to maintain + backwards capability. + To allow this, variables can now be either name value pairs (the current format): + <variable_name>: '<value_string>' + - or - + <variable_name>: {value: '<value_string>', report_events: [true|false]}

+

ARI: Added paths to get and set multiple channel variables.

+

Author: Ben Ford + Date: 2026-04-15

+

Two new paths exist for ARI to get and set multiple channel variables at + the same time. This is done via GET and POST like the single get and set + variable equivalents. Leading and trailing whitespace will be stripped + from the variable names for both paths. When setting variables, the + values will be read as-is, whitespace included. GET takes in a single + string with comma-separated values, while POST takes in a dictionary of + key value pairs. The code follows the same paths as when setting + multiple variables when originating a channel via ARI.

+

UserNote: Added new ARI paths for getting and setting multiple channel + variables at a time. For GET, this takes in a single string of + comma-separated variable names, while POST takes in a dictionary of key + value pairs. The behavior is the same as passing in variables when + originating a channel.

+

res_stir_shaken: avoid direct ASN1_STRING accesses

+

Author: Bernd Kuhls + Date: 2026-05-02

+

https://github.com/openssl/openssl/issues/29117

+

Signed-off-by: Bernd Kuhls bernd@kuhls.net

+

Resolves: #1952

+

tcptls.c: fix build with OpenSSL 4

+

Author: Bernd Kuhls + Date: 2026-05-02

+

tcptls.c: In function '__ssl_setup': + tcptls.c:417:52: error: implicit declaration of function 'SSLv3_client_method'; + did you mean 'SSLv23_client_method'? [-Wimplicit-function-declaration] + 417 | cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method());

+

SSLv3_client_method was removed from OpenSSL 4.0.0: + https://github.com/openssl/openssl/blob/openssl-4.0.0/doc/man7/ossl-removed-api.pod?plain=1#L440

+

Signed-off-by: Bernd Kuhls bernd@kuhls.net

+

Resolves: #1952

+

res_calendar: Fix build with libical 4.X

+

Author: mikhail_grishak + Date: 2026-05-26

+

libical 4.0 removed the icaltime_add() function in favor of icaltime_adjust(). Additionally, the callback signature for icalcomponent_foreach_recurrence() was updated to use a const pointer for the icaltime_span argument.

+

This commit adds conditional compilation using ICAL_MAJOR_VERSION to support both libical 3.X and the new 4.X API, ensuring backward compatibility.

+

Fixes: #1957

+

app_record: Fix hangup handling during beep playback

+

Author: UpBeta + Date: 2026-05-23

+

When a hangup occurs while app_record is playing the initial beep, + the application does not detect the hangup and continues running + until the maxduration timeout expires.

+

Replace the manual ast_streamfile() + ast_waitstream() sequence with + ast_stream_and_wait(), which properly detects hangup and returns + non-zero, allowing the application to exit immediately with + RECORD_STATUS set to HANGUP.

+

Resolves: #1950

+

odbc: Don't use prepared statements for distinct SQL statements

+

Author: smtcbn + Date: 2025-04-25

+

Avoids unnecessary prepare for simple INSERT statements that cause + issues with ProxySQL (prepared statement counter overflow).

+

Resolves: #1217

+

abstract_jb.c: Remove timerfd from channel when disabling jitter buffer

+

Author: Alexander Bakker + Date: 2026-05-20

+

Previously, the lingering timerfd would cause a tight loop if the + channel enters a BridgeWait after the jitter buffer was disabled.

+

Fixes: #1762

+

res_pjsip: Don't allow a leading period when wildcard matching

+

Author: Sean Bright + Date: 2026-05-26

+

The reference identifier (what the client provides - in this case a + hostname) must start with a domain label, not a ..

+

The current implementation will match .seanbright.com against + *.seanbright.com which is incorrect.

+

Ensure channel locks aren't held while calling ast_set_variables.

+

Author: George Joseph + Date: 2026-05-20

+

If the channel is locked when calling ast_set_variables and any of the + variables contained dialplan functions, there's a possiblilty of a deadlock. + To prevent this, either the explicit locks were removed or the call to + ast_set_variables moved out of the lock scope. A warning to not hold + channel locks is also added to the documentation for ast_set_variables.

+

Resolves: #1936

+

app_queue: fix double increment of member->calls with shared_lastcall

+

Author: smtcbn + Date: 2026-01-23

+

Under high concurrency, update_queue() may be invoked multiple times + for the same call, causing member->calls and queue-level counters to + be incremented more than once.

+

The existing starttime check is not atomic and allows concurrent + execution paths to pass. Treat member->starttime as a single-use token + and consume it via CAS to ensure the call is counted exactly once.

+

This also prevents incorrect call distribution when using strategies + such as fewestcalls.

+

Observed as a regression after upgrading to 20.17.

+

Resolves: #1736

+

chan_dahdi: Fix set but not used in mfcr2_show_links_of().

+

Author: George Joseph + Date: 2026-05-21

+

When openr2 is installed mfcr2_show_links_of() is no longer ifdeffed out + which makes gcc-16 complain with 'variable ‘x’ set but not used'.

+

Resolves: #1947

+

tests: add tests/test_codec_translations.c

+

Author: Sebastian Jennen + Date: 2026-03-06

+

This tests checks [slin -> codec -> slin] and then compares slin in vs out + regarding signal noise ratio and delay.

+

Near-lossless codecs (ulaw, alaw) are checked with a maximum per-sample + error bound. Lossy codecs are checked with a per-codec SNR threshold. + Cross-correlation alignment compensates for algorithmic delay in codecs + like speex and opus.

+

Covered codecs: ulaw, alaw, adpcm, g726, g726aal2, gsm, speex, + speex16, speex32, ilbc, codec2, lpc10, g722, opus.

+

Resolves: #1812

+

install_prereq: Add a 'minimal' mode for basic build dependencies

+

Author: Sean Bright + Date: 2026-05-20

+

chan_websocket: Handle incoming CONTINUATION frames.

+

Author: George Joseph + Date: 2026-05-20

+

chan_websocket now tells res_http_websocket to accumulate incoming CONTINUATION + frames into 1024 byte TEXT or BINARY frames.

+

Resolves: #1941

+

res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat().

+

Author: George Joseph + Date: 2026-05-19

+

AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ + AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ + rtp->rtcp->stdev_rxjitter);

+

Should have been

+

AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ + AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ + rtp->rtcp->stdev_rxmes);

+

Note the last macro parameter name.

+

Resolves: #1938

+

jansson: Upgrade version to jansson 2.15.0

+

Author: Stanislav Abramenkov + Date: 2026-05-13

+

UpgradeNote: jansson has been upgraded to 2.15.0. For more + information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0

+

Resolves: #1931

+

channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup.

+

Author: George Joseph + Date: 2026-05-12

+

The original trigger for setting the RTP stats in ast_softhangup() came from + an ARI issue where stats weren't being set in time to be reported on STASIS_END + events. The thought was that setting them in a common place like ast_softhangup() + would ensure the stats were set in possibly other scenarios. Unfortunately, + setting the RTP stats variables in ast_softhangup() broke ABI as it required + that no channel locks be held which was not the case earlier.

+

Given that the original issue was ARI, we can move setting the stats to + ast_ari_channels_hangup() in resource_channels just before it calls + ast_softhangup(). This might not catch all cases of the stats not being set, + but it won't break ABI or deadlock either.

+

Resolves: #1928

+

res_rtp_asterisk: Add option to control stun host resolution when TTL = 0

+

Author: George Joseph + Date: 2026-05-05

+

If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution + is enabled based on the TTL returned in the DNS results. If the TTL returned + is 0, it means that the next time the IP address is needed, it must be + looked up again. I.E. Don't cache. Historically (and incorrectly) however, + res_rtp_asterisk stopped the periodic resolution and never re-resolved the + hostname again.

+

Besides what's mentioned in the user notes... + * Additional debugging was added in various STUN/DNS functions. + * The rtp show settings CLI command shows more detailed STUN info. + * Some debugging was added to dns_core.c and dns_recurring.c.

+

UserNote: A new stunaddr_reresolve_ttl_0 parameter has been added to rtp.conf + that allows control over what happens when a STUN server hostname lookup + returns a TTL of 0. The values can be set as follows: + - 'no': This is the historical (and current default) behavior of not doing + any further lookups and continuing to use the last successful result until + Asterisk is restarted or rtp.conf is reloaded. + - 'yes': Use the last cached result for the current call but trigger + re-resolution in the background for the benefit of future calls. + If the result of the background lookup is a ttl > 0, periodic resolution + will be restarted otherwise the next call will use the new cached value + and will trigger a background lookup again.

+

UserNote: A new CLI command rtp resolve stun hostname has been added + that will force a resolution of the STUN hostname and (re)start periodic + resolution if the result has a TTL > 0.

+

Resolves: #1858

+

pjsip_configuration: Show actual dtls_verify config.

+

Author: Jaco Kroon + Date: 2026-05-07

+

Rather than merely showing

+

dtls_verify : Yes/No

+

in pjsip show endpoint xxx it will now be shown what exactly is being + checked, ie, one of:

+

dtls_verify : No + dtls_verify : Fingerprint + dtls_verify : Certificate + dtls_verify : Yes

+

Where Yes implies both Fingerprint and Certificate.

+

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

+

app_dial: Properly handle callee hangup while sending digits.

+

Author: Naveen Albert + Date: 2026-05-05

+

If we are sending digits (either DTMF, MF, or SF) to the called channel + after receiving progress or a wink, and the callee hangs up before we + have finished sending it digits, there are several problems that can ensue:

+ +

This is generally an edge case that occurs due to some kind of signaling + failure, but to better handle this:

+ +

Resolves: #1915

+

UserNote: If a called channel sends progress or wink and the caller begins + sending digits but the callee answers and then hangs up before digit + sending can finish, the call is now answered before being disconnected. + If the callee hangs up without answering, the call now continues in + the dialplan.

+

res_pjsip_messaging: Update To URI only if it is a SIP(S) URI

+

Author: Maximilian Fridrich + Date: 2026-05-07

+

When a message is sent via ARI, the ARI endpoint only provides a To + field which is also used as destination field. This means that the To + field might not necessarily contain a SIP URI but might instead specify + an Asterisk endpoint (in MessageDestinationInfo format). This led to + many warnings even though the message was sent correctly.

+

The fix is to only call ast_sip_update_to_uri if the To field starts + with the sip: or sips: scheme.

+

Resolves: #1357

+

Upgrade bundled pjproject to 2.17.

+

Author: Stanislav Abramenkov + Date: 2026-04-27

+

Resolves: #1888

+

UserNote: Bundled pjproject has been upgraded to 2.17. For more + information about what is included in this release, see the + pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17

+

res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug

+

Author: Mike Bradeen + Date: 2026-05-06

+

crypto_utils uses ast_asprintf to allocate the search string when checking the + certificate subject, but was not using ast_free to free it. This caused a crash + when Asterisk was built with malloc_debug

+

Resolves: #1921

+

manager: Eliminate unnecessary code, simplify sessions in stasis callbacks

+

Author: Joshua C. Colp + Date: 2026-05-04

+

Due to stasis filtering the stasis callback for AMI type messages is + guaranteed to only receive messages that can be turned into AMI events, + so remove the check done in the callback.

+

The sessions container usage for the stasis callbacks has also been + simplified by having a reference on the message router subscription + instead of having to acquire the sessions from the global object each + time.

+

res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup

+

Author: Peter Krall + Date: 2026-04-17

+

Modified the bridge playback teardown so the worker thread removes only the + playback control, while the after-bridge callback removes the playback + wrapper once the announcer has actually left the bridge.

+

This avoids a stale window where a new playback request could create a + replacement announcer before the old announcer had fully exited the holding + bridge.

+

Also replaced the flexible trailing bridge_id storage in the shared worker + thread data with an optional bridge_id pointer, since recording paths use the + same structure without a bridge id.

+

Fixes: #1861

+

res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage

+

Author: Sebastian Denz + Date: 2026-03-26

+

channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars

+

Author: George Joseph + Date: 2026-05-05

+

ast_softhangup() was locking the channel before calling ast_rtp_instance_set_stats_vars() + which, if the channel was in a bridge, then locked the bridge peer channel. If another + thread attempted to set bridge variables on the peer, it would lock that channel first, + then this channel causing a lock inversion. ast_softhangup() now holds the channel lock + while retrieving the rtp instance, then unlocks it before calling + ast_rtp_instance_set_stats_vars(), then locks it again after it returns.

+

Resolves: #1907

+

chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER

+

Author: Charles Langlois + Date: 2026-04-16

+

When a PJSIP endpoint is configured with set_var invoking a dialplan + function (e.g. PJSIP_HEADER(add,...)), chan_pjsip_new() calls + pbx_builtin_setvar_helper() while holding the channel lock. + For function-style variables, this dispatches to ast_func_write() + which, in the case of PJSIP_HEADER, calls + ast_sip_push_task_wait_serializer() -- blocking synchronously while + the channel lock is held.

+

If a concurrent operation (ARI, AMI, rtp_check_timeout) traverses + the channels container via ast_channel_get_by_name(), it acquires + the container lock then tries to lock individual channels in the + iteration callback (by_uniqueid_cb/by_name_cb). When the serializer + thread also needs the container lock, a circular dependency forms:

+
channel_lock -> serializer_wait -> container_lock -> channel_lock
+
+

This causes a complete Asterisk freeze. In the observed case, 36 + threads were blocked on the container lock until res_freeze_check + triggered SIGABRT after its 30-second timeout.

+

Unlock the channel before iterating endpoint channel_vars so that + dialplan functions can block without holding the channel lock. Re-lock + the channel for ast_channel_stage_snapshot_done() so the batched + snapshot is published under lock and captures the full channel state + including the variables set during the loop.

+

Fixes: #1872

+

res_pjsip: Add per-endpoint RTP port range configuration

+

Author: mattia + Date: 2026-04-01

+

Add rtp_port_start and rtp_port_end options to PJSIP endpoint + configuration, allowing each endpoint to use a dedicated RTP port + range instead of the global rtp.conf setting.

+

This is useful for scenarios where different endpoints need isolated + port ranges, such as firewall rules per trunk, multi-tenant systems, + or network QoS policies tied to port ranges.

+

The implementation adds ast_rtp_instance_new_with_port_range() to the + RTP engine API, which sets the port range on the instance before the + engine allocates the transport. The default RTP engine + (res_rtp_asterisk) checks for per-instance overrides in + rtp_allocate_transport() and falls back to the global range when + none is set.

+

Both options must be set together, with values >= 1024 and + rtp_port_end > rtp_port_start. Setting both to 0 (the default) + preserves existing behavior.

+

Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71

+

UserNote: PJSIP endpoints now support rtp_port_start and + rtp_port_end options to configure a dedicated RTP port range per + endpoint, overriding the global rtp.conf setting.

+

UpgradeNote: An alembic database migration has been added to add + the rtp_port_start and rtp_port_end columns to the ps_endpoints + table. Run "alembic upgrade head" to apply the schema change.

+

DeveloperNote: New public API: ast_rtp_instance_new_with_port_range() + creates an RTP instance with a per-instance port range. + ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() + allow RTP engines to query the override. Third-party RTP engines can + use these getters to support per-instance port ranges.

+

app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules

+

Author: phoneben + Date: 2026-04-26

+

app_queue: Fix raise_respect_min not copied in copy_rules() causing rN rules to be ignored.

+

copy_rules() never copied raise_respect_min into the per-call rule list, so the flag was always 0 when a timed penaltychange rule fired, making rN behave like plain N and raising members below min_penalty that should have been excluded.

+

Also fixes update_qe_rule() not propagating the flag from qe->pr to qe, and dropping the r prefix when saving back to QUEUE_RAISE_PENALTY.

+

Resolves: #1901

+

app_voicemail_odbc: fix msgnum race and crash on failed STORE

+

Author: phoneben + Date: 2026-04-09

+

app_voicemail_odbc: fix msgnum race and crash on failed STORE

+

Two concurrent callers leaving voicemail to the same mailbox could be + assigned the same msgnum because ast_unlock_path() was called before + STORE(), allowing a second thread to read the same LAST_MSG_INDEX() + before the first INSERT committed. The losing thread got a duplicate + key error, but execution continued into notify_new_message() -> + RETRIEVE() because the STORE() return value was not checked. + RETRIEVE() then fetched the winning thread's DB row, mmap'd its blob + size against the locally truncated file, and crashed with SIGBUS.

+

Hold the path lock through STORE() and bail out on failure.

+

Fixes: #1653

+

ari_websockets: Fix two issues in the cleanup of outbound websockets.

+

Author: George Joseph + Date: 2026-04-22

+
    +
  1. +

    session_cleanup() now saves the websocket type before unlinking the + session from the session registry. This prevents a FRACK when cleaning + up per-call websockets when MALLOC_DEBUG is used.

    +
    2. +
  2. +

    session_shutdown_cb() and outbound_sessions_load() now call + pthread_cancel() to cancel the session handler thread to prevent the + thread from continually trying to connect to a server after the + connection config has been removed by a reload. This required the + thread to use pthread_cleanup_push() to clean up its reference to the + session instead of RAII because RAII destructors don't get run when + pthread_cancel() is used.

    +
    3. +
+

Resolves: #1894

+

compat.h: Ensure check for __STDC_VERSION__ is not attempted for c++.

+

Author: George Joseph + Date: 2026-04-27

+

__STDC_VERSION__ is specific to C but up until gcc 16, the g++ compiler + also defined it. With g++ 16.0 it's no longer defined (which is the correct + behavior) so compiling channelstorage_cpp_map_name_id.cc fails. The + check for __STDC_VERSION__ in compat.h is now skipped if we're compiling + a C++ source file.

+

Resolves: #1903

+

pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c

+

Author: phoneben + Date: 2026-04-22

+

Backport pjsip/pjproject#4941 which fixes a build/link failure when + compiling against OpenSSL < 1.1.0 (e.g. OpenSSL 1.0.2k on CentOS 7).

+

Two symbols introduced in OpenSSL 1.1.x were called unconditionally + in ssl_sock_ossl.c without version guards:

+ +

Without this fix, linking fails with: + undefined reference to TLS_method' + undefined reference toSSL_CTX_set_ciphersuites'

+

when building Asterisk with bundled pjproject on systems such as + CentOS 7 with OpenSSL 1.0.2k.

+

Resolves: #1892

+

asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE.

+

Author: George Joseph + Date: 2026-04-22

+

Line 2729 has #if HAVE_LIBEDIT_IS_UNICODE instead if #ifdef. Since + macros defined by autoconf are either set to 1 or not set at all, + older distros where libedit isn't unicode won't have that macro defined + and will fail to compile.

+

Resolves: #1896

+

cdrel_custom: fix SQLite compatibility for versions < 3.20.0

+

Author: phoneben + Date: 2026-04-21

+

cdrel_custom: fix SQLite compatibility for versions < 3.20.0

+

Replace sqlite3_prepare_v3 + SQLITE_PREPARE_PERSISTENT with a version-guarded fallback to sqlite3_prepare_v2 for older SQLite builds.

+

Resolves: #1885

+

translate.c: implement different sample_types for translation computation.

+

Author: Sebastian Jennen + Date: 2026-04-02

+

The default (codec) still uses the codec provided samples. Additionally + different sample_types can be used with eg: translate sampletype speech + and then running core show translation comp 10 to measure performance + of different audio scenarios.

+

Resolves: #1807

+

stasis_broadcast: Add optional ARI broadcast with first-claim-wins

+

Author: Daniel Donoghue + Date: 2026-02-25

+

Adds two optional modules: + res_stasis_broadcast.so: Infrastructure for broadcasting a single incoming + channel to multiple ARI applications with atomic first-claim-wins semantics.

+

app_stasis_broadcast.so: Provides the StasisBroadcast() dialplan application + which invokes the broadcast infrastructure.

+

Both modules are self-contained; if neither is loaded there is zero runtime + impact. Loading them does not alter existing Stasis or ARI behavior unless + explicitly used.

+

Key Features (only active when modules are loaded): + Fisher-Yates shuffled broadcast dispatch for fair claim races + Atomic claim operations using mutex + condition variable signaling + Configurable broadcast timeouts + Safe regex application filtering with validation to mitigate ReDoS risk + Thread-safe channel variable snapshotting (channel locked during reads) + Late-claim safety: broadcast context kept alive until after the Stasis + session ends so concurrent claimants always receive 409 Conflict rather + than 404 Not Found + Memory safety via RAII_VAR, ast_json_ref/unref, and ao2 reference counting

+

Components Added: + res/res_stasis_broadcast.c: Core broadcast + claim logic + apps/app_stasis_broadcast.c: StasisBroadcast() dialplan application + include/asterisk/stasis_app_broadcast.h: Public API header + res/ari/resource_events.c: Integrates POST /ari/events/claim endpoint + rest-api/api-docs/events.json: New CallBroadcast and CallClaimed events

+

Implementation Notes: + Broadcast contexts reside in an ao2 hash container keyed by channel id. Each + context holds atomic claim state, winner application name, timeout metadata, + and a condition variable for waiters. Broadcast contexts are kept alive until + after stasis_app_exec() returns so that concurrent claimants racing against + the timeout always receive 409 Conflict. Broadcast dispatch calls + stasis_app_send() directly for each matching application in shuffled order. + Regex filters are validated with bounded length, group depth, quantified + group count, and alternation limits to reduce pathological backtracking. + Timeout calculation uses timespec arithmetic with overflow-safe millisecond + remainder handling. Event JSON follows existing Stasis/ARI conventions; + references are managed correctly to avoid leaks or double frees.

+

Optional Nature / Impact: + No changes to existing APIs, events, or applications when absent. + Clean fallback: systems ignoring the modules behave identically to prior + versions.

+

Development was assisted by Claude (Anthropic). All generated code has been + reviewed, tested, and is understood by the author.

+

UserNote: New optional modules res_stasis_broadcast.so and + app_stasis_broadcast.so enable broadcasting an incoming channel to multiple + ARI applications. The first application to successfully claim (via + POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan + application initiates broadcasts. CallBroadcast and CallClaimed events notify + applications. When modules are not loaded, behavior is unchanged.

+

DeveloperNote: New public APIs in stasis_app_broadcast.h: + stasis_app_broadcast_channel(), stasis_app_claim_channel(), + stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event + types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; + no existing ABI modified.

+

res_audiosocket: Tolerate non-audio frame types

+

Author: Sven Kube + Date: 2026-04-22

+

This commit implements the handling of non-voice or DTMF frames like the + chan_websocket handling added in #1588. Rather than treating unsupported + frames as fatal errors, silently ignore CNG frames and log a warning for + other unsupported types.

+

pbx_functions: Save module pointer before calling read and write callbacks.

+

Author: George Joseph + Date: 2026-04-21

+

Before ast_func_read and ast_func_write call their respective read and write + callbacks for registered dialplan functions, they use the module pointer in + the registered ast_custom_function structure to increment the module use + count. They then decrement the usecount when the callback returns. This + prevents the providing module from being unloaded while there's a call using + the function.

+

Some modules, notably func_odbc, create and destroy dialplan functions based + on the contents of a config file. Since the ast_custom_function structure is + dynamically allocated, it could be destroyed on reload which means when the + module's read or write callback returns to the ast_func calls it would try to + decrement the usecount using the module pointer from an ast_custom_function + structure that has already been freed. Proper locking or reference counting + by the module can reduce the possibility of this happening but it can't + prevent it because it doesn't have control after its read or write callback + has returned to ast_func_read or ast_func_write.

+

To address this, ast_func_read, ast_func_read2 and ast_func_write save the + module pointer to a local variable before calling the module's callback, + then use the saved pointer to decrement the use count. The module + pointer will always be valid if the module is loaded regardless of the + state of the ast_custom_function structure.

+

Resolves: #1818

+

chan_iax2: Add CHANNEL getter to retrieve auth method.

+

Author: Naveen Albert + Date: 2026-04-18

+

Add a property to the CHANNEL method to retrieve the auth method, + which can be used to retrieve the specific auth method actually + negotiated for a call (e.g. RSA, MD5, etc.).

+

Also clean up some of the documentation for the secure properties + to clarify how these relate to call encryption.

+

Resolves: #1878

+

UserNote: CHANNEL(auth_method) can now be used to retrieve the + auth method negotiated for a call on IAX2 channels.

+

fix: backport pjproject stdatomic.h GCC 4.8 build failure patch

+

Author: phoneben + Date: 2026-04-21

+

pjproject 2.16 (bundled) fails to build on GCC 4.8 (CentOS/RHEL 7) + due to a false positive C11 atomics detection introduced in pjproject + commit #4570. A fix has been submitted upstream to pjproject (#4933).

+

Adding a local patch to third-party/pjproject/patches/ until a fixed + version of pjproject is bundled in Asterisk.

+

Fixes build error: + ../src/pj/os_core_unix.c:52:27: fatal error: stdatomic.h: No such file or directory

+

Resolves: #1883

+

res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy.

+

Author: George Joseph + Date: 2026-04-16

+

The rtp_ioqueue_thread_destroy() function was destroying the the ioqueue + thread and releasing its pool but not destroying the ioqueue itself. This + was causing the ioqueue's epoll file descriptor to leak.

+

Resolves: #1867

+

res_pjsip_maintenance: Add PJSIP endpoint maintenance mode

+

Author: Daniel Donoghue + Date: 2026-03-10

+

Introduces res_pjsip_maintenance, a loadable module that allows + operators to place individual PJSIP endpoints into maintenance mode + at runtime without unregistering or disabling them.

+

While an endpoint is in maintenance mode: + * New inbound INVITE and SUBSCRIBE dialogs are rejected with + 503 Service Unavailable and a Retry-After: 300 header. + * In-progress dialogs (re-INVITE, UPDATE, BYE, etc.) are + unaffected and complete normally. + * Outbound originations via Dial() or ARI originate are refused + before any SIP session is created.

+

State is held in-memory only and is cleared on module unload + or Asterisk restart.

+

This module was developed with AI assistance (Claude). All code + has been reviewed and tested by the author, who takes full + responsibility for the submission.

+

CLI interface: + pjsip set maintenance + pjsip show maintenance [endpoint]

+

AMI interface: + Action: PJSIPSetMaintenance + Endpoint: |all + State: on|off

+
Action: PJSIPShowMaintenance
+Endpoint: <name>  (optional; omit to list all)
+
+Emits PJSIPMaintenanceStatus events per result, followed by
+PJSIPMaintenanceStatusComplete. State changes also emit an
+unsolicited PJSIPMaintenanceStatus event.
+
+

To support outbound blocking, a new session_create callback is + added to ast_sip_session_supplement. Supplements that set this + callback are invoked at the start of ast_sip_session_create_outgoing() + in res_pjsip_session, before any dialog or invite session resources + are allocated. res_pjsip_maintenance registers itself as a session + supplement and uses this callback to gate outbound session creation + on a per-endpoint basis.

+

MODULEINFO: + pjproject + res_pjsip + res_pjsip_session

+

UserNote: New module res_pjsip_maintenance adds runtime maintenance + mode for PJSIP endpoints. Use "pjsip set maintenance + " to enable or disable, and "pjsip show maintenance" + to list affected endpoints. AMI actions PJSIPSetMaintenance and + PJSIPShowMaintenance provide programmatic access. No configuration + file changes required.

+

DeveloperNote: ast_sip_session_supplement gains a new optional + callback - int (session_create)(struct ast_sip_endpoint endpoint, + const char *destination). It is called from the global supplement + list (not per-session) at the start of ast_sip_session_create_outgoing() + via ast_sip_session_check_supplement_create(). Returning non-zero + blocks the outgoing session. Modules that need to gate outbound + SIP session creation should register a supplement with this callback + set rather than hooking into chan_pjsip directly.

+

chan_iax2: Add another check to abort frame handling if datalen < 0.

+

Author: Naveen Albert + Date: 2026-04-11

+

Commit 2da221e217cbff957af928e8df43ee25583232d1 added a missing abort + if datalen < 0 check on a code path and an assertion inside + iax_frame_wrap if we ever encountered a frame with a negative frame + length (which will eventually cause a crash).

+

Add another missing abort check for negative datalen, exposed by this + assertion. (Similar to the previous commit, this is a video frame with + a datalen of -1).

+

Resolves: #1865

+

res_pjsip_outbound_registration: only update the Expires header if the value has changed

+

Author: Mike Bradeen + Date: 2026-04-08

+

The PJSIP outbound registration API has undocumented behavior when reconfiguring + the outbound registration if the expires value being set is the same as what was + previously set.

+

In this case PJSIP will remove the Expires header entirely from subsequent + outbound REGISTER requests. To eliminate this as an issue we now check the current + expires value against the configured expires value and only apply it if it differs.

+

This ensures that outbound REGISTER requests always contain an Expires header.

+

Resolves: #1859

+

func_talkdetect.c: Clarify dsp_talking_threshold documentation.

+

Author: Sean Bright + Date: 2026-04-08

+

Fixes: #1761

+

make_xml_documentation: Remove temporary file on script exit.

+

Author: Sean Bright + Date: 2026-04-09

+

Fixes: #1862

+

res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread

+

Author: George Joseph + Date: 2026-04-07

+

When res_pjsip is reloaded directly, it does the sorcery reload in a pjsip + servant thread as it's supposed to. res_pjsip_config_wizard however + was not which was leading to occasional deadlocks. It now does the reload + in a servant thread just like res_pjsip.

+

Resolves: #1855

+

build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build

+

Author: Alexei Gradinari + Date: 2026-04-06

+

The pjsua Python module and the pjsua/pjsystest apps were used by the + Asterisk Test Suite for SIP simulation in dev mode builds. They are now + fully obsolete for three independent reasons:

+
    +
  1. +

    pjsua Python bindings officially deprecated upstream. The pjproject + maintainers added pjsip-apps/src/python/DEPRECATED.txt directing + users to the PJSUA2 SWIG binding instead. A build-fix PR + (https://github.com/pjsip/pjproject/pull/4892) was closed by the + maintainer explicitly citing this deprecation.

    +
    2. +
  2. +

    Removed from the Asterisk Test Suite. As confirmed by @mbradeen: + > "We had to get rid of pjsua when we went to Python3 because it would + > hang due to a conflict between async calls within pjsua and twisted. + > There are still some old references to tests we couldn't fully convert + > to sipp, but those are skipped."

    +
    3. +
  3. +

    Broken and unmaintained. Building with Python 2.7 (the only version + configure.ac searched for) fails with: + _pjsua.c: error: 'INIT_RETURN' undeclared (first use in this function) + due to a bug in pjproject 2.16's python3_compat.h that upstream + declined to fix.

    +
    4. +
+

This PR removes all pjsua-related build artifacts from Asterisk's bundled + pjproject build: the pjsua and pjsystest application binaries, the deprecated + Python (_pjsua.so) bindings, the asterisk_malloc_debug.c stubs, and the + PYTHONDEV detection from configure.ac. Also removes libpjsua from + Asterisk's main linker flags.

+

DeveloperNote: The pjsua and pjsystest application binaries, the deprecated + Python pjsua bindings (_pjsua.so), and the asterisk_malloc_debug.c stub + implementations are no longer built or installed as part of the bundled + pjproject dev mode build. The PYTHONDEV (python2.7-dev) build dependency + is also removed. Developers who relied on the pjsua binary for Test Suite + SIP simulation should use SIPp instead, which is the current Asterisk Test + Suite standard.

+

Fixes: #1840

+

callerid: fix signed char causing crash in MDMF parser

+

Author: Milan Kyselica + Date: 2026-03-25

+

Change rawdata buffer from char to unsigned char to prevent + sign-extension of TLV length bytes >= 0x80. On signed-char + platforms (all Asterisk builds due to -fsigned-char in + configure.ac), these values become negative when assigned to + int, bypass the if (res > 32) bounds check, and reach + memcpy as size_t producing a ~18 EB read that immediately + crashes with SIGSEGV.

+

Affects DAHDI analog (FXO) channels only. Not reachable + via SIP, PRI/BRI, or DTMF-based Caller ID.

+

Fixes: #1839

+ diff --git a/ChangeLogs/ChangeLog-22.10.0-rc1.md b/ChangeLogs/ChangeLog-22.10.0-rc1.md new file mode 100644 index 0000000000..5e25c6e447 --- /dev/null +++ b/ChangeLogs/ChangeLog-22.10.0-rc1.md @@ -0,0 +1,1336 @@ + +## Change Log for Release asterisk-22.10.0-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 53 +- Commit Authors: 24 +- Issues Resolved: 43 +- Security Advisories Resolved: 0 + +### User Notes: + +- #### res ari: Add attachable states to Channels and Bridges + Bridge variables now can be set and retrieved via the following paths: + `/bridges/{bridgeId}/variable` + `/bridges/{bridgeId}/variables` + Both Bridge and Channel variables can now be set with an optional 'report_events' + boolean flag that will cause those variables to be included on all events on that + object. The 'report_events' flag will default to False if not set to maintain + backwards capability. + To allow this, variables can now be either name value pairs (the current format): + `: ''` + - or - + `: {value: '', report_events: [true|false]}` + +- #### ARI: Added paths to get and set multiple channel variables. + Added new ARI paths for getting and setting multiple channel + variables at a time. For GET, this takes in a single string of + comma-separated variable names, while POST takes in a dictionary of key + value pairs. The behavior is the same as passing in variables when + originating a channel. + +- #### res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 + A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf + that allows control over what happens when a STUN server hostname lookup + returns a TTL of 0. The values can be set as follows: + - 'no': This is the historical (and current default) behavior of not doing + any further lookups and continuing to use the last successful result until + Asterisk is restarted or rtp.conf is reloaded. + - 'yes': Use the last cached result for the current call but trigger + re-resolution in the background for the benefit of future calls. + If the result of the background lookup is a ttl > 0, periodic resolution + will be restarted otherwise the next call will use the new cached value + and will trigger a background lookup again. + A new CLI command `rtp resolve stun hostname` has been added +- #### app_dial: Properly handle callee hangup while sending digits. + If a called channel sends progress or wink and the caller begins + sending digits but the callee answers and then hangs up before digit + sending can finish, the call is now answered before being disconnected. + If the callee hangs up without answering, the call now continues in + the dialplan. + +- #### Upgrade bundled pjproject to 2.17. + Bundled pjproject has been upgraded to 2.17. For more + information about what is included in this release, see the + pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17 + +- #### res_pjsip: Add per-endpoint RTP port range configuration + PJSIP endpoints now support rtp_port_start and + rtp_port_end options to configure a dedicated RTP port range per + endpoint, overriding the global rtp.conf setting. + +- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins + New optional modules res_stasis_broadcast.so and + app_stasis_broadcast.so enable broadcasting an incoming channel to multiple + ARI applications. The first application to successfully claim (via + POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan + application initiates broadcasts. CallBroadcast and CallClaimed events notify + applications. When modules are not loaded, behavior is unchanged. + +- #### chan_iax2: Add CHANNEL getter to retrieve auth method. + CHANNEL(auth_method) can now be used to retrieve the + auth method negotiated for a call on IAX2 channels. + +- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode + New module res_pjsip_maintenance adds runtime maintenance + mode for PJSIP endpoints. Use "pjsip set maintenance + " to enable or disable, and "pjsip show maintenance" + to list affected endpoints. AMI actions PJSIPSetMaintenance and + PJSIPShowMaintenance provide programmatic access. No configuration + file changes required. + + +### Upgrade Notes: + +- #### jansson: Upgrade version to jansson 2.15.0 + jansson has been upgraded to 2.15.0. For more + information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0 + +- #### res_pjsip: Add per-endpoint RTP port range configuration + An alembic database migration has been added to add + the rtp_port_start and rtp_port_end columns to the ps_endpoints + table. Run "alembic upgrade head" to apply the schema change. + + +### Developer Notes: + +- #### res_pjsip: Add per-endpoint RTP port range configuration + New public API: ast_rtp_instance_new_with_port_range() + creates an RTP instance with a per-instance port range. + ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() + allow RTP engines to query the override. Third-party RTP engines can + use these getters to support per-instance port ranges. + +- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins + New public APIs in stasis_app_broadcast.h: + stasis_app_broadcast_channel(), stasis_app_claim_channel(), + stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event + types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; + no existing ABI modified. + +- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode + ast_sip_session_supplement gains a new optional + callback - int (*session_create)(struct ast_sip_endpoint *endpoint, + const char *destination). It is called from the global supplement + list (not per-session) at the start of ast_sip_session_create_outgoing() + via ast_sip_session_check_supplement_create(). Returning non-zero + blocks the outgoing session. Modules that need to gate outbound + SIP session creation should register a supplement with this callback + set rather than hooking into chan_pjsip directly. + +- #### build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build + The pjsua and pjsystest application binaries, the deprecated + Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub + implementations are no longer built or installed as part of the bundled + pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency + is also removed. Developers who relied on the pjsua binary for Test Suite + SIP simulation should use SIPp instead, which is the current Asterisk Test + Suite standard. + Fixes: #1840 + + +### Commit Authors: + +- Alexander Bakker: (1) +- Alexei Gradinari: (1) +- Ben Ford: (1) +- Bernd Kuhls: (2) +- Charles Langlois: (1) +- Daniel Donoghue: (2) +- George Joseph: (14) +- Jaco Kroon: (1) +- Joshua C. Colp: (1) +- Maximilian Fridrich: (1) +- Mike Bradeen: (3) +- Milan Kyselica: (1) +- Naveen Albert: (3) +- Peter Krall: (1) +- Sean Bright: (4) +- Sebastian Denz: (1) +- Sebastian Jennen: (2) +- Stanislav Abramenkov: (2) +- Sven Kube: (1) +- UpBeta: (1) +- mattia: (1) +- mikhail_grishak: (1) +- phoneben: (5) +- smtcbn: (2) + +## Issue and Commit Detail: + +### Closed Issues: + + - 1217: [bug]: INSERT INTO cdr query prepare statement issue on cdr_adaptive_odbc to control statement preparation manually + - 1357: [bug]: MessageSend WARNING “not a valid SIP/SIPS URI” when using endpoint not URI + - 1653: [bug]: Asterisk ODBC Voicemail Crash Caused by Voicemail Re-entry Loop and Unsafe BLOB Retrieval + - 1736: app_queue: update_queue() may double-increment member->calls with shared_lastcall=yes (regression observed after 20.17; impacts fewestcalls routing) + - 1761: func_talkdetect.c: TALK_DETECT docs wording mistake + - 1762: [bug]: 100% CPU usage when entering BridgeWait after JITTERBUFFER(disabled)= + - 1807: [new-feature]: translate.c: implement different types of sample frame inputs + - 1812: [new-feature]: add tests/test_codec_translations.c + - 1818: [bug]: func_odbc: possible use-after-free crash during reload with active calls + - 1839: Crash in MDMF Caller ID parser due to signed char length field on DAHDI channels + - 1840: [bug]: Asterisk fails to compile with --enable-dev-mode=yes due to INIT_RETURN undeclared in bundled pjproject Python bindings + - 1855: [bug]: core reload deadlocks Asterisk (pjsip, CLI, etc.) + - 1858: [bug]: DNS records with a TTL of zero are permanently cached + - 1859: [bug]: res_pjsip_outbound_registration: No expires header set when triggered via CLI + - 1861: [bug]: Possible heap corruption in audiohook/translate write path during bridged media + - 1862: [bug]: Build fails with Building Documentation: line 210: /tmp/xmldoc.tmp.xml: Permission denied + - 1865: [bug]: chan_iax2: Another code path that causes crashes on negative data lengths + - 1867: [bug]: Massive [eventpoll] file-descriptor leak (hundreds of epoll fds) when TURN is enabled in rtp.conf + - 1872: [bug]: Deadlock in chan_pjsip_new when endpoint set_var invokes PJSIP_HEADER + - 1878: [new-feature]: chan_iax2: Allow retrieving the auth method using the CHANNEL function + - 1883: [bug]: fix: stdatomic.h false positive on GCC 4.8 + - 1885: [bug]: cdrel_custom :SQLite version too old: sqlite3_prepare_v3 / SQLITE_PREPARE_PERSISTENT undeclared + - 1888: [improvement]: pjsip: Upgrade bundled version to pjproject 2.17 + - 1892: [bug]: Build failure with bundled pjproject on OpenSSL 1.0.x: undefined reference to TLS_method and SSL_CTX_set_ciphersuites + - 1894: [bug]: Outbound ARI websockets don't always clean up completely + - 1896: [bug]: asterisk.c fails to compile when HAVE_LIBEDIT_IS_UNICODE isn't defined + - 1901: [bug]: QUEUE_RAISE_PENALTY=rN ignored when set via queue rules + - 1903: [bug]: g++ 16 no longer defines __STDC_VERSION__ causing channelstorage_cpp_map_name_id.cc to fail + - 1907: [bug]: Deadlock between bridge and setting of RTP stats variables at hangup + - 1910: [improvement]: Add attachable state variables to Channels and Bridges. + - 1915: [bug]: app_dial: Channel not handled properly if callee disconnects while caller is sending it digits prior to answer + - 1921: [bug]: Memory error in crypto_get_cert_subject when using malloc_debug + - 1928: [bug]: Calling ast_softhangup with channel lock held can cause deadlock + - 1931: [improvement]: jansson: Upgrade version to jansson 2.15.0 + - 1936: [bug]: Calling set_variable on PJSIP channel when originating with ARI with PJSIP_HEADER can result in deadlock + - 1938: [bug]: res_rtp_asterisk: Copy/paste error in ast_rtp_get_stat() + - 1941: [bug]: chan_websocket doesn't handle CONTINUATION websocket frames + - 1947: [bug]: chan_dahdi fails to build with gcc-16 when openr2 is installed + - 1950: [bug]: app_record does not detect channel hangup during beep playback + - 1952: [bug]: OpenSSL 4.0.0 + - 1957: [bug]: Calendar module fails to build with libical 4.X + - 1970: [bug]: Startup or shutdown segfault in res_ari_model under certain conditions with DEVMODE and persistent outbound websockets. + +### Commits By Author: + +- #### Alexander Bakker (1): + - abstract_jb.c: Remove timerfd from channel when disabling jitter buffer + +- #### Alexei Gradinari (1): + - build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build + +- #### Ben Ford (1): + - ARI: Added paths to get and set multiple channel variables. + +- #### Bernd Kuhls (2): + - res_stir_shaken: avoid direct ASN1_STRING accesses + - tcptls.c: fix build with OpenSSL 4 + +- #### Charles Langlois (1): + - chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER + +- #### Daniel Donoghue (2): + - stasis_broadcast: Add optional ARI broadcast with first-claim-wins + - res_pjsip_maintenance: Add PJSIP endpoint maintenance mode + +- #### George Joseph (14): + - res_ari: Add res_ari_model as an optional_module. + - Ensure channel locks aren't held while calling ast_set_variables. + - chan_dahdi: Fix set but not used in mfcr2_show_links_of(). + - chan_websocket: Handle incoming CONTINUATION frames. + - res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). + - channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. + - res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 + - channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars + - ari_websockets: Fix two issues in the cleanup of outbound websockets. + - compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++. + - asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. + - pbx_functions: Save module pointer before calling read and write callbacks. + - res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy. + - res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread + +- #### Jaco Kroon (1): + - pjsip_configuration: Show actual dtls_verify config. + +- #### Joshua C. Colp (1): + - manager: Eliminate unnecessary code, simplify sessions in stasis callbacks + +- #### Maximilian Fridrich (1): + - res_pjsip_messaging: Update To URI only if it is a SIP(S) URI + +- #### Mike Bradeen (3): + - res ari: Add attachable states to Channels and Bridges + - res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug + - res_pjsip_outbound_registration: only update the Expires header if the value has changed + +- #### Milan Kyselica (1): + - callerid: fix signed char causing crash in MDMF parser + +- #### Naveen Albert (3): + - app_dial: Properly handle callee hangup while sending digits. + - chan_iax2: Add CHANNEL getter to retrieve auth method. + - chan_iax2: Add another check to abort frame handling if datalen < 0. + +- #### Peter Krall (1): + - res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup + +- #### Sean Bright (4): + - res_pjsip: Don't allow a leading period when wildcard matching + - install_prereq: Add a 'minimal' mode for basic build dependencies + - func_talkdetect.c: Clarify dsp_talking_threshold documentation. + - make_xml_documentation: Remove temporary file on script exit. + +- #### Sebastian Denz (1): + - res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage + +- #### Sebastian Jennen (2): + - tests: add tests/test_codec_translations.c + - translate.c: implement different sample_types for translation computation. + +- #### Stanislav Abramenkov (2): + - jansson: Upgrade version to jansson 2.15.0 + - Upgrade bundled pjproject to 2.17. + +- #### Sven Kube (1): + - res_audiosocket: Tolerate non-audio frame types + +- #### UpBeta (1): + - app_record: Fix hangup handling during beep playback + +- #### mattia (1): + - res_pjsip: Add per-endpoint RTP port range configuration + +- #### mikhail_grishak (1): + - res_calendar: Fix build with libical 4.X + +- #### phoneben (5): + - app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules + - app_voicemail_odbc: fix msgnum race and crash on failed STORE + - pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c + - cdrel_custom: fix SQLite compatibility for versions < 3.20.0 + - fix: backport pjproject stdatomic.h GCC 4.8 build failure patch + +- #### smtcbn (2): + - odbc: Don't use prepared statements for distinct SQL statements + - app_queue: fix double increment of member->calls with shared_lastcall + +### Commit List: + +- res_ari: Add res_ari_model as an optional_module. +- res ari: Add attachable states to Channels and Bridges +- ARI: Added paths to get and set multiple channel variables. +- res_stir_shaken: avoid direct ASN1_STRING accesses +- tcptls.c: fix build with OpenSSL 4 +- res_calendar: Fix build with libical 4.X +- app_record: Fix hangup handling during beep playback +- odbc: Don't use prepared statements for distinct SQL statements +- abstract_jb.c: Remove timerfd from channel when disabling jitter buffer +- res_pjsip: Don't allow a leading period when wildcard matching +- Ensure channel locks aren't held while calling ast_set_variables. +- app_queue: fix double increment of member->calls with shared_lastcall +- chan_dahdi: Fix set but not used in mfcr2_show_links_of(). +- tests: add tests/test_codec_translations.c +- install_prereq: Add a 'minimal' mode for basic build dependencies +- chan_websocket: Handle incoming CONTINUATION frames. +- res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). +- jansson: Upgrade version to jansson 2.15.0 +- channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. +- res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 +- pjsip_configuration: Show actual dtls_verify config. +- app_dial: Properly handle callee hangup while sending digits. +- res_pjsip_messaging: Update To URI only if it is a SIP(S) URI +- Upgrade bundled pjproject to 2.17. +- res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug +- manager: Eliminate unnecessary code, simplify sessions in stasis callbacks +- res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup +- res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage +- channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars +- chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER +- res_pjsip: Add per-endpoint RTP port range configuration +- app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules +- app_voicemail_odbc: fix msgnum race and crash on failed STORE +- ari_websockets: Fix two issues in the cleanup of outbound websockets. +- compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++. +- pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c +- asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. +- cdrel_custom: fix SQLite compatibility for versions < 3.20.0 +- translate.c: implement different sample_types for translation computation. +- stasis_broadcast: Add optional ARI broadcast with first-claim-wins +- res_audiosocket: Tolerate non-audio frame types +- pbx_functions: Save module pointer before calling read and write callbacks. +- chan_iax2: Add CHANNEL getter to retrieve auth method. +- fix: backport pjproject stdatomic.h GCC 4.8 build failure patch +- res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy. +- res_pjsip_maintenance: Add PJSIP endpoint maintenance mode +- chan_iax2: Add another check to abort frame handling if datalen < 0. +- res_pjsip_outbound_registration: only update the Expires header if the value has changed +- func_talkdetect.c: Clarify dsp_talking_threshold documentation. +- make_xml_documentation: Remove temporary file on script exit. +- res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread +- build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build +- callerid: fix signed char causing crash in MDMF parser + +### Commit Details: + +#### res_ari: Add res_ari_model as an optional_module. + Author: George Joseph + Date: 2026-06-03 + + Under certain timing/load conditions, res_ari_model may not load until after + res_ari on startup or it might unload before res_ari on shutdown. This can + cause a segfault when DEVMODE is enabled and there are persistent outbound + websocket connections because DEVMODE forces validation of outgoing events + against the models. To prevent this, res_ari_model has been added as an + "optional_module" to res_ari's NODULE_INFO. This will enforce load/unload + order but not make res_ari dependent on res_ari_model. However, if + Asterisk is configured with --enable-dev-mode, res_ari will fail to + load if res_ari_model isn't available. + + Resolves: #1970 + +#### res ari: Add attachable states to Channels and Bridges + Author: Mike Bradeen + Date: 2026-03-31 + + Adds the ability to attach multiple states to both Channels and Bridges in the form + of variables that are included in all events on the associated object. + + First, this adds an optional boolean field to channel variables 'report_events' + that causes the variable to automatically be included in all events on that channel. + + To allow this, variables can now be either name value pairs (the current format): + `: ''` + - or - + `: {value: '', report_events: [true|false]}` + + If the old format is used or 'report_events' is not included, it will default to + false and retain current behavior. + + Second, this extends both reported and unreported variables to Bridges so they too + may have stateful information. + + Resolves: #1910 + + UserNote: Bridge variables now can be set and retrieved via the following paths: + `/bridges/{bridgeId}/variable` + `/bridges/{bridgeId}/variables` + Both Bridge and Channel variables can now be set with an optional 'report_events' + boolean flag that will cause those variables to be included on all events on that + object. The 'report_events' flag will default to False if not set to maintain + backwards capability. + To allow this, variables can now be either name value pairs (the current format): + `: ''` + - or - + `: {value: '', report_events: [true|false]}` + +#### ARI: Added paths to get and set multiple channel variables. + Author: Ben Ford + Date: 2026-04-15 + + Two new paths exist for ARI to get and set multiple channel variables at + the same time. This is done via GET and POST like the single get and set + variable equivalents. Leading and trailing whitespace will be stripped + from the variable names for both paths. When setting variables, the + values will be read as-is, whitespace included. GET takes in a single + string with comma-separated values, while POST takes in a dictionary of + key value pairs. The code follows the same paths as when setting + multiple variables when originating a channel via ARI. + + UserNote: Added new ARI paths for getting and setting multiple channel + variables at a time. For GET, this takes in a single string of + comma-separated variable names, while POST takes in a dictionary of key + value pairs. The behavior is the same as passing in variables when + originating a channel. + +#### res_stir_shaken: avoid direct ASN1_STRING accesses + Author: Bernd Kuhls + Date: 2026-05-02 + + https://github.com/openssl/openssl/issues/29117 + + Signed-off-by: Bernd Kuhls + + Resolves: #1952 + +#### tcptls.c: fix build with OpenSSL 4 + Author: Bernd Kuhls + Date: 2026-05-02 + + tcptls.c: In function '__ssl_setup': + tcptls.c:417:52: error: implicit declaration of function 'SSLv3_client_method'; + did you mean 'SSLv23_client_method'? [-Wimplicit-function-declaration] + 417 | cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method()); + + SSLv3_client_method was removed from OpenSSL 4.0.0: + https://github.com/openssl/openssl/blob/openssl-4.0.0/doc/man7/ossl-removed-api.pod?plain=1#L440 + + Signed-off-by: Bernd Kuhls + + Resolves: #1952 + +#### res_calendar: Fix build with libical 4.X + Author: mikhail_grishak + Date: 2026-05-26 + + libical 4.0 removed the icaltime_add() function in favor of icaltime_adjust(). Additionally, the callback signature for icalcomponent_foreach_recurrence() was updated to use a const pointer for the icaltime_span argument. + + This commit adds conditional compilation using ICAL_MAJOR_VERSION to support both libical 3.X and the new 4.X API, ensuring backward compatibility. + + Fixes: #1957 + +#### app_record: Fix hangup handling during beep playback + Author: UpBeta + Date: 2026-05-23 + + When a hangup occurs while app_record is playing the initial beep, + the application does not detect the hangup and continues running + until the maxduration timeout expires. + + Replace the manual ast_streamfile() + ast_waitstream() sequence with + ast_stream_and_wait(), which properly detects hangup and returns + non-zero, allowing the application to exit immediately with + RECORD_STATUS set to HANGUP. + + Resolves: #1950 + +#### odbc: Don't use prepared statements for distinct SQL statements + Author: smtcbn + Date: 2025-04-25 + + Avoids unnecessary prepare for simple INSERT statements that cause + issues with ProxySQL (prepared statement counter overflow). + + Resolves: #1217 + +#### abstract_jb.c: Remove timerfd from channel when disabling jitter buffer + Author: Alexander Bakker + Date: 2026-05-20 + + Previously, the lingering timerfd would cause a tight loop if the + channel enters a BridgeWait after the jitter buffer was disabled. + + Fixes: #1762 + +#### res_pjsip: Don't allow a leading period when wildcard matching + Author: Sean Bright + Date: 2026-05-26 + + The reference identifier (what the client provides - in this case a + hostname) must start with a domain label, not a `.`. + + The current implementation will match `.seanbright.com` against + `*.seanbright.com` which is incorrect. + +#### Ensure channel locks aren't held while calling ast_set_variables. + Author: George Joseph + Date: 2026-05-20 + + If the channel is locked when calling ast_set_variables and any of the + variables contained dialplan functions, there's a possiblilty of a deadlock. + To prevent this, either the explicit locks were removed or the call to + ast_set_variables moved out of the lock scope. A warning to not hold + channel locks is also added to the documentation for ast_set_variables. + + Resolves: #1936 + +#### app_queue: fix double increment of member->calls with shared_lastcall + Author: smtcbn + Date: 2026-01-23 + + Under high concurrency, update_queue() may be invoked multiple times + for the same call, causing member->calls and queue-level counters to + be incremented more than once. + + The existing starttime check is not atomic and allows concurrent + execution paths to pass. Treat member->starttime as a single-use token + and consume it via CAS to ensure the call is counted exactly once. + + This also prevents incorrect call distribution when using strategies + such as fewestcalls. + + Observed as a regression after upgrading to 20.17. + + Resolves: #1736 + +#### chan_dahdi: Fix set but not used in mfcr2_show_links_of(). + Author: George Joseph + Date: 2026-05-21 + + When openr2 is installed mfcr2_show_links_of() is no longer ifdeffed out + which makes gcc-16 complain with 'variable ‘x’ set but not used'. + + Resolves: #1947 + +#### tests: add tests/test_codec_translations.c + Author: Sebastian Jennen + Date: 2026-03-06 + + This tests checks [slin -> codec -> slin] and then compares slin in vs out + regarding signal noise ratio and delay. + + Near-lossless codecs (ulaw, alaw) are checked with a maximum per-sample + error bound. Lossy codecs are checked with a per-codec SNR threshold. + Cross-correlation alignment compensates for algorithmic delay in codecs + like speex and opus. + + Covered codecs: ulaw, alaw, adpcm, g726, g726aal2, gsm, speex, + speex16, speex32, ilbc, codec2, lpc10, g722, opus. + + Resolves: #1812 + +#### install_prereq: Add a 'minimal' mode for basic build dependencies + Author: Sean Bright + Date: 2026-05-20 + + +#### chan_websocket: Handle incoming CONTINUATION frames. + Author: George Joseph + Date: 2026-05-20 + + chan_websocket now tells res_http_websocket to accumulate incoming CONTINUATION + frames into 1024 byte TEXT or BINARY frames. + + Resolves: #1941 + +#### res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). + Author: George Joseph + Date: 2026-05-19 + + ``` + AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ + AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ + rtp->rtcp->stdev_rxjitter); + ``` + + Should have been + + ``` + AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ + AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ + rtp->rtcp->stdev_rxmes); + ``` + + Note the last macro parameter name. + + Resolves: #1938 + +#### jansson: Upgrade version to jansson 2.15.0 + Author: Stanislav Abramenkov + Date: 2026-05-13 + + UpgradeNote: jansson has been upgraded to 2.15.0. For more + information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0 + + Resolves: #1931 + +#### channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. + Author: George Joseph + Date: 2026-05-12 + + The original trigger for setting the RTP stats in ast_softhangup() came from + an ARI issue where stats weren't being set in time to be reported on STASIS_END + events. The thought was that setting them in a common place like ast_softhangup() + would ensure the stats were set in possibly other scenarios. Unfortunately, + setting the RTP stats variables in ast_softhangup() broke ABI as it required + that no channel locks be held which was not the case earlier. + + Given that the original issue was ARI, we can move setting the stats to + ast_ari_channels_hangup() in resource_channels just before it calls + ast_softhangup(). This might not catch all cases of the stats not being set, + but it won't break ABI or deadlock either. + + Resolves: #1928 + +#### res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 + Author: George Joseph + Date: 2026-05-05 + + If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution + is enabled based on the TTL returned in the DNS results. If the TTL returned + is 0, it means that the next time the IP address is needed, it must be + looked up again. I.E. Don't cache. Historically (and incorrectly) however, + res_rtp_asterisk stopped the periodic resolution and never re-resolved the + hostname again. + + Besides what's mentioned in the user notes... + * Additional debugging was added in various STUN/DNS functions. + * The `rtp show settings` CLI command shows more detailed STUN info. + * Some debugging was added to dns_core.c and dns_recurring.c. + + UserNote: A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf + that allows control over what happens when a STUN server hostname lookup + returns a TTL of 0. The values can be set as follows: + - 'no': This is the historical (and current default) behavior of not doing + any further lookups and continuing to use the last successful result until + Asterisk is restarted or rtp.conf is reloaded. + - 'yes': Use the last cached result for the current call but trigger + re-resolution in the background for the benefit of future calls. + If the result of the background lookup is a ttl > 0, periodic resolution + will be restarted otherwise the next call will use the new cached value + and will trigger a background lookup again. + + UserNote: A new CLI command `rtp resolve stun hostname` has been added + that will force a resolution of the STUN hostname and (re)start periodic + resolution if the result has a TTL > 0. + + Resolves: #1858 + +#### pjsip_configuration: Show actual dtls_verify config. + Author: Jaco Kroon + Date: 2026-05-07 + + Rather than merely showing + + dtls_verify : Yes/No + + in pjsip show endpoint xxx it will now be shown what exactly is being + checked, ie, one of: + + dtls_verify : No + dtls_verify : Fingerprint + dtls_verify : Certificate + dtls_verify : Yes + + Where Yes implies both Fingerprint and Certificate. + + Signed-off-by: Jaco Kroon + +#### app_dial: Properly handle callee hangup while sending digits. + Author: Naveen Albert + Date: 2026-05-05 + + If we are sending digits (either DTMF, MF, or SF) to the called channel + after receiving progress or a wink, and the callee hangs up before we + have finished sending it digits, there are several problems that can ensue: + + * If the callee hung up without answering, the calling channel would + hang up and not continue in the dialplan. + * If the callee *did* answer before hanging up, the answer was never + passed through to the caller, since this gets "eaten" by the various + digit streaming functions and is never processed by app_dial. + + This is generally an edge case that occurs due to some kind of signaling + failure, but to better handle this: + + * Set to_answer to 0 to prevent hangup on the exit path, just like other + parts of wait_for_answer. + * Better document this usage of to_answer. + * If the channel did answer while it was receiving digits, manually + answer the calling channel before we abort. The call would not continue + in the dialplan anyways (either before or after this fix), but technically + the call was answered, so the CDRs should probably reflect that, and this + mirrors the behavior of calls which normally do not continue. + + Resolves: #1915 + + UserNote: If a called channel sends progress or wink and the caller begins + sending digits but the callee answers and then hangs up before digit + sending can finish, the call is now answered before being disconnected. + If the callee hangs up without answering, the call now continues in + the dialplan. + +#### res_pjsip_messaging: Update To URI only if it is a SIP(S) URI + Author: Maximilian Fridrich + Date: 2026-05-07 + + When a message is sent via ARI, the ARI endpoint only provides a To + field which is also used as destination field. This means that the To + field might not necessarily contain a SIP URI but might instead specify + an Asterisk endpoint (in MessageDestinationInfo format). This led to + many warnings even though the message was sent correctly. + + The fix is to only call `ast_sip_update_to_uri` if the To field starts + with the sip: or sips: scheme. + + Resolves: #1357 + +#### Upgrade bundled pjproject to 2.17. + Author: Stanislav Abramenkov + Date: 2026-04-27 + + Resolves: #1888 + + UserNote: Bundled pjproject has been upgraded to 2.17. For more + information about what is included in this release, see the + pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17 + +#### res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug + Author: Mike Bradeen + Date: 2026-05-06 + + crypto_utils uses ast_asprintf to allocate the search string when checking the + certificate subject, but was not using ast_free to free it. This caused a crash + when Asterisk was built with malloc_debug + + Resolves: #1921 + +#### manager: Eliminate unnecessary code, simplify sessions in stasis callbacks + Author: Joshua C. Colp + Date: 2026-05-04 + + Due to stasis filtering the stasis callback for AMI type messages is + guaranteed to only receive messages that can be turned into AMI events, + so remove the check done in the callback. + + The sessions container usage for the stasis callbacks has also been + simplified by having a reference on the message router subscription + instead of having to acquire the sessions from the global object each + time. + +#### res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup + Author: Peter Krall + Date: 2026-04-17 + + Modified the bridge playback teardown so the worker thread removes only the + playback control, while the after-bridge callback removes the playback + wrapper once the announcer has actually left the bridge. + + This avoids a stale window where a new playback request could create a + replacement announcer before the old announcer had fully exited the holding + bridge. + + Also replaced the flexible trailing bridge_id storage in the shared worker + thread data with an optional bridge_id pointer, since recording paths use the + same structure without a bridge id. + + Fixes: #1861 + +#### res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage + Author: Sebastian Denz + Date: 2026-03-26 + + +#### channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars + Author: George Joseph + Date: 2026-05-05 + + ast_softhangup() was locking the channel before calling ast_rtp_instance_set_stats_vars() + which, if the channel was in a bridge, then locked the bridge peer channel. If another + thread attempted to set bridge variables on the peer, it would lock that channel first, + then this channel causing a lock inversion. ast_softhangup() now holds the channel lock + while retrieving the rtp instance, then unlocks it before calling + ast_rtp_instance_set_stats_vars(), then locks it again after it returns. + + Resolves: #1907 + +#### chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER + Author: Charles Langlois + Date: 2026-04-16 + + When a PJSIP endpoint is configured with set_var invoking a dialplan + function (e.g. PJSIP_HEADER(add,...)), chan_pjsip_new() calls + pbx_builtin_setvar_helper() while holding the channel lock. + For function-style variables, this dispatches to ast_func_write() + which, in the case of PJSIP_HEADER, calls + ast_sip_push_task_wait_serializer() -- blocking synchronously while + the channel lock is held. + + If a concurrent operation (ARI, AMI, rtp_check_timeout) traverses + the channels container via ast_channel_get_by_name(), it acquires + the container lock then tries to lock individual channels in the + iteration callback (by_uniqueid_cb/by_name_cb). When the serializer + thread also needs the container lock, a circular dependency forms: + + channel_lock -> serializer_wait -> container_lock -> channel_lock + + This causes a complete Asterisk freeze. In the observed case, 36 + threads were blocked on the container lock until res_freeze_check + triggered SIGABRT after its 30-second timeout. + + Unlock the channel before iterating endpoint channel_vars so that + dialplan functions can block without holding the channel lock. Re-lock + the channel for ast_channel_stage_snapshot_done() so the batched + snapshot is published under lock and captures the full channel state + including the variables set during the loop. + + Fixes: #1872 + +#### res_pjsip: Add per-endpoint RTP port range configuration + Author: mattia + Date: 2026-04-01 + + Add rtp_port_start and rtp_port_end options to PJSIP endpoint + configuration, allowing each endpoint to use a dedicated RTP port + range instead of the global rtp.conf setting. + + This is useful for scenarios where different endpoints need isolated + port ranges, such as firewall rules per trunk, multi-tenant systems, + or network QoS policies tied to port ranges. + + The implementation adds ast_rtp_instance_new_with_port_range() to the + RTP engine API, which sets the port range on the instance before the + engine allocates the transport. The default RTP engine + (res_rtp_asterisk) checks for per-instance overrides in + rtp_allocate_transport() and falls back to the global range when + none is set. + + Both options must be set together, with values >= 1024 and + rtp_port_end > rtp_port_start. Setting both to 0 (the default) + preserves existing behavior. + + Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71 + + UserNote: PJSIP endpoints now support rtp_port_start and + rtp_port_end options to configure a dedicated RTP port range per + endpoint, overriding the global rtp.conf setting. + + UpgradeNote: An alembic database migration has been added to add + the rtp_port_start and rtp_port_end columns to the ps_endpoints + table. Run "alembic upgrade head" to apply the schema change. + + DeveloperNote: New public API: ast_rtp_instance_new_with_port_range() + creates an RTP instance with a per-instance port range. + ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() + allow RTP engines to query the override. Third-party RTP engines can + use these getters to support per-instance port ranges. + +#### app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules + Author: phoneben + Date: 2026-04-26 + + app_queue: Fix raise_respect_min not copied in copy_rules() causing rN rules to be ignored. + + `copy_rules()` never copied `raise_respect_min` into the per-call rule list, so the flag was always 0 when a timed penaltychange rule fired, making `rN` behave like plain `N` and raising members below `min_penalty` that should have been excluded. + + Also fixes `update_qe_rule()` not propagating the flag from `qe->pr` to `qe`, and dropping the `r` prefix when saving back to `QUEUE_RAISE_PENALTY`. + + Resolves: #1901 + +#### app_voicemail_odbc: fix msgnum race and crash on failed STORE + Author: phoneben + Date: 2026-04-09 + + app_voicemail_odbc: fix msgnum race and crash on failed STORE + + Two concurrent callers leaving voicemail to the same mailbox could be + assigned the same msgnum because ast_unlock_path() was called before + STORE(), allowing a second thread to read the same LAST_MSG_INDEX() + before the first INSERT committed. The losing thread got a duplicate + key error, but execution continued into notify_new_message() -> + RETRIEVE() because the STORE() return value was not checked. + RETRIEVE() then fetched the winning thread's DB row, mmap'd its blob + size against the locally truncated file, and crashed with SIGBUS. + + Hold the path lock through STORE() and bail out on failure. + + Fixes: #1653 + +#### ari_websockets: Fix two issues in the cleanup of outbound websockets. + Author: George Joseph + Date: 2026-04-22 + + 1. session_cleanup() now saves the websocket type before unlinking the + session from the session registry. This prevents a FRACK when cleaning + up per-call websockets when MALLOC_DEBUG is used. + + 2. session_shutdown_cb() and outbound_sessions_load() now call + pthread_cancel() to cancel the session handler thread to prevent the + thread from continually trying to connect to a server after the + connection config has been removed by a reload. This required the + thread to use pthread_cleanup_push() to clean up its reference to the + session instead of RAII because RAII destructors don't get run when + pthread_cancel() is used. + + Resolves: #1894 + +#### compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++. + Author: George Joseph + Date: 2026-04-27 + + `__STDC_VERSION__` is specific to C but up until gcc 16, the g++ compiler + also defined it. With g++ 16.0 it's no longer defined (which is the correct + behavior) so compiling channelstorage_cpp_map_name_id.cc fails. The + check for `__STDC_VERSION__` in compat.h is now skipped if we're compiling + a C++ source file. + + Resolves: #1903 + +#### pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c + Author: phoneben + Date: 2026-04-22 + + Backport pjsip/pjproject#4941 which fixes a build/link failure when + compiling against OpenSSL < 1.1.0 (e.g. OpenSSL 1.0.2k on CentOS 7). + + Two symbols introduced in OpenSSL 1.1.x were called unconditionally + in ssl_sock_ossl.c without version guards: + + - `TLS_method()` in `init_ossl_ctx()` is now guarded with + `OPENSSL_VERSION_NUMBER < 0x10100000L`, falling back to + `SSLv23_method()` on older OpenSSL. + + - `SSL_CTX_set_ciphersuites()` is now guarded with + `OPENSSL_VERSION_NUMBER >= 0x1010100fL` since this function + was introduced in OpenSSL 1.1.1 and is absent in 1.0.x. + + Without this fix, linking fails with: + undefined reference to `TLS_method' + undefined reference to `SSL_CTX_set_ciphersuites' + + when building Asterisk with bundled pjproject on systems such as + CentOS 7 with OpenSSL 1.0.2k. + + Resolves: #1892 + +#### asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. + Author: George Joseph + Date: 2026-04-22 + + Line 2729 has `#if HAVE_LIBEDIT_IS_UNICODE` instead if `#ifdef`. Since + macros defined by autoconf are either set to `1` or not set at all, + older distros where libedit isn't unicode won't have that macro defined + and will fail to compile. + + Resolves: #1896 + +#### cdrel_custom: fix SQLite compatibility for versions < 3.20.0 + Author: phoneben + Date: 2026-04-21 + + cdrel_custom: fix SQLite compatibility for versions < 3.20.0 + + Replace sqlite3_prepare_v3 + SQLITE_PREPARE_PERSISTENT with a version-guarded fallback to sqlite3_prepare_v2 for older SQLite builds. + + Resolves: #1885 + +#### translate.c: implement different sample_types for translation computation. + Author: Sebastian Jennen + Date: 2026-04-02 + + The default (codec) still uses the codec provided samples. Additionally + different sample_types can be used with eg: `translate sampletype speech` + and then running `core show translation comp 10` to measure performance + of different audio scenarios. + + Resolves: #1807 + +#### stasis_broadcast: Add optional ARI broadcast with first-claim-wins + Author: Daniel Donoghue + Date: 2026-02-25 + + Adds two optional modules: + res_stasis_broadcast.so: Infrastructure for broadcasting a single incoming + channel to multiple ARI applications with atomic first-claim-wins semantics. + + app_stasis_broadcast.so: Provides the StasisBroadcast() dialplan application + which invokes the broadcast infrastructure. + + Both modules are self-contained; if neither is loaded there is zero runtime + impact. Loading them does not alter existing Stasis or ARI behavior unless + explicitly used. + + Key Features (only active when modules are loaded): + Fisher-Yates shuffled broadcast dispatch for fair claim races + Atomic claim operations using mutex + condition variable signaling + Configurable broadcast timeouts + Safe regex application filtering with validation to mitigate ReDoS risk + Thread-safe channel variable snapshotting (channel locked during reads) + Late-claim safety: broadcast context kept alive until after the Stasis + session ends so concurrent claimants always receive 409 Conflict rather + than 404 Not Found + Memory safety via RAII_VAR, ast_json_ref/unref, and ao2 reference counting + + Components Added: + res/res_stasis_broadcast.c: Core broadcast + claim logic + apps/app_stasis_broadcast.c: StasisBroadcast() dialplan application + include/asterisk/stasis_app_broadcast.h: Public API header + res/ari/resource_events.c: Integrates POST /ari/events/claim endpoint + rest-api/api-docs/events.json: New CallBroadcast and CallClaimed events + + Implementation Notes: + Broadcast contexts reside in an ao2 hash container keyed by channel id. Each + context holds atomic claim state, winner application name, timeout metadata, + and a condition variable for waiters. Broadcast contexts are kept alive until + after stasis_app_exec() returns so that concurrent claimants racing against + the timeout always receive 409 Conflict. Broadcast dispatch calls + stasis_app_send() directly for each matching application in shuffled order. + Regex filters are validated with bounded length, group depth, quantified + group count, and alternation limits to reduce pathological backtracking. + Timeout calculation uses timespec arithmetic with overflow-safe millisecond + remainder handling. Event JSON follows existing Stasis/ARI conventions; + references are managed correctly to avoid leaks or double frees. + + Optional Nature / Impact: + No changes to existing APIs, events, or applications when absent. + Clean fallback: systems ignoring the modules behave identically to prior + versions. + + Development was assisted by Claude (Anthropic). All generated code has been + reviewed, tested, and is understood by the author. + + UserNote: New optional modules res_stasis_broadcast.so and + app_stasis_broadcast.so enable broadcasting an incoming channel to multiple + ARI applications. The first application to successfully claim (via + POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan + application initiates broadcasts. CallBroadcast and CallClaimed events notify + applications. When modules are not loaded, behavior is unchanged. + + DeveloperNote: New public APIs in stasis_app_broadcast.h: + stasis_app_broadcast_channel(), stasis_app_claim_channel(), + stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event + types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; + no existing ABI modified. + +#### res_audiosocket: Tolerate non-audio frame types + Author: Sven Kube + Date: 2026-04-22 + + This commit implements the handling of non-voice or DTMF frames like the + chan_websocket handling added in #1588. Rather than treating unsupported + frames as fatal errors, silently ignore CNG frames and log a warning for + other unsupported types. + +#### pbx_functions: Save module pointer before calling read and write callbacks. + Author: George Joseph + Date: 2026-04-21 + + Before ast_func_read and ast_func_write call their respective read and write + callbacks for registered dialplan functions, they use the module pointer in + the registered ast_custom_function structure to increment the module use + count. They then decrement the usecount when the callback returns. This + prevents the providing module from being unloaded while there's a call using + the function. + + Some modules, notably func_odbc, create and destroy dialplan functions based + on the contents of a config file. Since the ast_custom_function structure is + dynamically allocated, it could be destroyed on reload which means when the + module's read or write callback returns to the ast_func calls it would try to + decrement the usecount using the module pointer from an ast_custom_function + structure that has already been freed. Proper locking or reference counting + by the module can reduce the possibility of this happening but it can't + prevent it because it doesn't have control after its read or write callback + has returned to ast_func_read or ast_func_write. + + To address this, ast_func_read, ast_func_read2 and ast_func_write save the + module pointer to a local variable before calling the module's callback, + then use the saved pointer to decrement the use count. The module + pointer will always be valid if the module is loaded regardless of the + state of the ast_custom_function structure. + + Resolves: #1818 + +#### chan_iax2: Add CHANNEL getter to retrieve auth method. + Author: Naveen Albert + Date: 2026-04-18 + + Add a property to the CHANNEL method to retrieve the auth method, + which can be used to retrieve the specific auth method actually + negotiated for a call (e.g. RSA, MD5, etc.). + + Also clean up some of the documentation for the secure properties + to clarify how these relate to call encryption. + + Resolves: #1878 + + UserNote: CHANNEL(auth_method) can now be used to retrieve the + auth method negotiated for a call on IAX2 channels. + +#### fix: backport pjproject stdatomic.h GCC 4.8 build failure patch + Author: phoneben + Date: 2026-04-21 + + pjproject 2.16 (bundled) fails to build on GCC 4.8 (CentOS/RHEL 7) + due to a false positive C11 atomics detection introduced in pjproject + commit #4570. A fix has been submitted upstream to pjproject (#4933). + + Adding a local patch to third-party/pjproject/patches/ until a fixed + version of pjproject is bundled in Asterisk. + + Fixes build error: + ../src/pj/os_core_unix.c:52:27: fatal error: stdatomic.h: No such file or directory + + Resolves: #1883 + +#### res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy. + Author: George Joseph + Date: 2026-04-16 + + The rtp_ioqueue_thread_destroy() function was destroying the the ioqueue + thread and releasing its pool but not destroying the ioqueue itself. This + was causing the ioqueue's epoll file descriptor to leak. + + Resolves: #1867 + +#### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode + Author: Daniel Donoghue + Date: 2026-03-10 + + Introduces res_pjsip_maintenance, a loadable module that allows + operators to place individual PJSIP endpoints into maintenance mode + at runtime without unregistering or disabling them. + + While an endpoint is in maintenance mode: + * New inbound INVITE and SUBSCRIBE dialogs are rejected with + 503 Service Unavailable and a Retry-After: 300 header. + * In-progress dialogs (re-INVITE, UPDATE, BYE, etc.) are + unaffected and complete normally. + * Outbound originations via Dial() or ARI originate are refused + before any SIP session is created. + + State is held in-memory only and is cleared on module unload + or Asterisk restart. + + This module was developed with AI assistance (Claude). All code + has been reviewed and tested by the author, who takes full + responsibility for the submission. + + CLI interface: + pjsip set maintenance + pjsip show maintenance [endpoint] + + AMI interface: + Action: PJSIPSetMaintenance + Endpoint: |all + State: on|off + + Action: PJSIPShowMaintenance + Endpoint: (optional; omit to list all) + + Emits PJSIPMaintenanceStatus events per result, followed by + PJSIPMaintenanceStatusComplete. State changes also emit an + unsolicited PJSIPMaintenanceStatus event. + + To support outbound blocking, a new session_create callback is + added to ast_sip_session_supplement. Supplements that set this + callback are invoked at the start of ast_sip_session_create_outgoing() + in res_pjsip_session, before any dialog or invite session resources + are allocated. res_pjsip_maintenance registers itself as a session + supplement and uses this callback to gate outbound session creation + on a per-endpoint basis. + + MODULEINFO: + pjproject + res_pjsip + res_pjsip_session + + UserNote: New module res_pjsip_maintenance adds runtime maintenance + mode for PJSIP endpoints. Use "pjsip set maintenance + " to enable or disable, and "pjsip show maintenance" + to list affected endpoints. AMI actions PJSIPSetMaintenance and + PJSIPShowMaintenance provide programmatic access. No configuration + file changes required. + + DeveloperNote: ast_sip_session_supplement gains a new optional + callback - int (*session_create)(struct ast_sip_endpoint *endpoint, + const char *destination). It is called from the global supplement + list (not per-session) at the start of ast_sip_session_create_outgoing() + via ast_sip_session_check_supplement_create(). Returning non-zero + blocks the outgoing session. Modules that need to gate outbound + SIP session creation should register a supplement with this callback + set rather than hooking into chan_pjsip directly. + +#### chan_iax2: Add another check to abort frame handling if datalen < 0. + Author: Naveen Albert + Date: 2026-04-11 + + Commit 2da221e217cbff957af928e8df43ee25583232d1 added a missing abort + if datalen < 0 check on a code path and an assertion inside + iax_frame_wrap if we ever encountered a frame with a negative frame + length (which will eventually cause a crash). + + Add another missing abort check for negative datalen, exposed by this + assertion. (Similar to the previous commit, this is a video frame with + a datalen of -1). + + Resolves: #1865 + +#### res_pjsip_outbound_registration: only update the Expires header if the value has changed + Author: Mike Bradeen + Date: 2026-04-08 + + The PJSIP outbound registration API has undocumented behavior when reconfiguring + the outbound registration if the expires value being set is the same as what was + previously set. + + In this case PJSIP will remove the Expires header entirely from subsequent + outbound REGISTER requests. To eliminate this as an issue we now check the current + expires value against the configured expires value and only apply it if it differs. + + This ensures that outbound REGISTER requests always contain an Expires header. + + Resolves: #1859 + +#### func_talkdetect.c: Clarify dsp_talking_threshold documentation. + Author: Sean Bright + Date: 2026-04-08 + + Fixes: #1761 + +#### make_xml_documentation: Remove temporary file on script exit. + Author: Sean Bright + Date: 2026-04-09 + + Fixes: #1862 + +#### res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread + Author: George Joseph + Date: 2026-04-07 + + When res_pjsip is reloaded directly, it does the sorcery reload in a pjsip + servant thread as it's supposed to. res_pjsip_config_wizard however + was not which was leading to occasional deadlocks. It now does the reload + in a servant thread just like res_pjsip. + + Resolves: #1855 + +#### build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build + Author: Alexei Gradinari + Date: 2026-04-06 + + The pjsua Python module and the pjsua/pjsystest apps were used by the + Asterisk Test Suite for SIP simulation in dev mode builds. They are now + fully obsolete for three independent reasons: + + 1. **pjsua Python bindings officially deprecated upstream.** The pjproject + maintainers added `pjsip-apps/src/python/DEPRECATED.txt` directing + users to the PJSUA2 SWIG binding instead. A build-fix PR + (https://github.com/pjsip/pjproject/pull/4892) was closed by the + maintainer explicitly citing this deprecation. + + 2. **Removed from the Asterisk Test Suite.** As confirmed by @mbradeen: + > *"We had to get rid of pjsua when we went to Python3 because it would + > hang due to a conflict between async calls within pjsua and twisted. + > There are still some old references to tests we couldn't fully convert + > to sipp, but those are skipped."* + + 3. **Broken and unmaintained.** Building with Python 2.7 (the only version + `configure.ac` searched for) fails with: + ``` + _pjsua.c: error: 'INIT_RETURN' undeclared (first use in this function) + ``` + due to a bug in pjproject 2.16's `python3_compat.h` that upstream + declined to fix. + + This PR removes all pjsua-related build artifacts from Asterisk's bundled + pjproject build: the pjsua and pjsystest application binaries, the deprecated + Python (`_pjsua.so`) bindings, the `asterisk_malloc_debug.c` stubs, and the + `PYTHONDEV` detection from `configure.ac`. Also removes `libpjsua` from + Asterisk's main linker flags. + + DeveloperNote: The pjsua and pjsystest application binaries, the deprecated + Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub + implementations are no longer built or installed as part of the bundled + pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency + is also removed. Developers who relied on the pjsua binary for Test Suite + SIP simulation should use SIPp instead, which is the current Asterisk Test + Suite standard. + + Fixes: #1840 + +#### callerid: fix signed char causing crash in MDMF parser + Author: Milan Kyselica + Date: 2026-03-25 + + Change rawdata buffer from char to unsigned char to prevent + sign-extension of TLV length bytes >= 0x80. On signed-char + platforms (all Asterisk builds due to -fsigned-char in + configure.ac), these values become negative when assigned to + int, bypass the `if (res > 32)` bounds check, and reach + memcpy as size_t producing a ~18 EB read that immediately + crashes with SIGSEGV. + + Affects DAHDI analog (FXO) channels only. Not reachable + via SIP, PRI/BRI, or DTMF-based Caller ID. + + Fixes: #1839 + diff --git a/README.html b/README.html index 530f5b2ecd..6446670bfd 100644 --- a/README.html +++ b/README.html @@ -1,4 +1,4 @@ -Readme for asterisk-22.9.0 +Readme for asterisk-22.10.0-rc1

The Asterisk(R) Open Source PBX

By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
 Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.

 
If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.

 
-
Change Logs

+
Change Logs

 
 
 
NEW INSTALLATIONS

diff --git a/README.md b/README.md
index 5a3841b272..f7a6727404 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.
 
 
-[Change Logs](ChangeLogs/ChangeLog-22.9.0.html)
+[Change Logs](ChangeLogs/ChangeLog-22.10.0-rc1.html)
 
 
 ### NEW INSTALLATIONS
diff --git a/contrib/realtime/mysql/mysql_config.sql b/contrib/realtime/mysql/mysql_config.sql
index 0c44c5c434..11ec93c38c 100644
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
@@ -1755,3 +1755,11 @@ ALTER TABLE ps_endpoints ADD COLUMN follow_redirect_methods VARCHAR(95);
 
 UPDATE alembic_version SET version_num='bb6d54e22913' WHERE alembic_version.version_num = 'dc7c357dc178';
 
+-- Running upgrade bb6d54e22913 -> e89e30cee53f
+
+ALTER TABLE ps_endpoints ADD COLUMN rtp_port_start INTEGER;
+
+ALTER TABLE ps_endpoints ADD COLUMN rtp_port_end INTEGER;
+
+UPDATE alembic_version SET version_num='e89e30cee53f' WHERE alembic_version.version_num = 'bb6d54e22913';
+
diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql
index 25324a9886..e4e26299dd 100644
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
@@ -1879,5 +1879,13 @@ ALTER TABLE ps_endpoints ADD COLUMN follow_redirect_methods VARCHAR(95);
 
 UPDATE alembic_version SET version_num='bb6d54e22913' WHERE alembic_version.version_num = 'dc7c357dc178';
 
+-- Running upgrade bb6d54e22913 -> e89e30cee53f
+
+ALTER TABLE ps_endpoints ADD COLUMN rtp_port_start INTEGER;
+
+ALTER TABLE ps_endpoints ADD COLUMN rtp_port_end INTEGER;
+
+UPDATE alembic_version SET version_num='e89e30cee53f' WHERE alembic_version.version_num = 'bb6d54e22913';
+
 COMMIT;