From bc8fc9549669b0de4634d9e74415b390d95ec7ee Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Mon, 23 Feb 2026 18:58:02 +0000 Subject: [PATCH] Update for certified-22.8-cert1-rc1 --- .version | 2 +- CHANGES.html | 2 +- CHANGES.md | 2 +- .../ChangeLog-certified-22.8-cert1-rc1.html | 13186 +++++++++++++ .../ChangeLog-certified-22.8-cert1-rc1.md | 15527 ++++++++++++++++ README.html | 4 +- README.md | 2 +- 7 files changed, 28719 insertions(+), 6 deletions(-) create mode 100644 ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.html create mode 100644 ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.md diff --git a/.version b/.version index 955174f9da..7a798a4efd 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -22.8.2 +certified-22.8-cert1-rc1 diff --git a/CHANGES.html b/CHANGES.html index 905be0926e..f022497268 120000 --- a/CHANGES.html +++ b/CHANGES.html @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.8.2.html \ No newline at end of file +ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index 6c73404570..bffb8fb387 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.8.2.md \ No newline at end of file +ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.html b/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.html new file mode 100644 index 0000000000..d506ed19de --- /dev/null +++ b/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.html @@ -0,0 +1,13186 @@ +ChangeLog for asterisk-certified-22.8-cert1-rc1 +

Change Log for Release asterisk-certified-22.8-cert1-rc1

+

Links:

+ +

Summary:

+ +

User Notes:

+ +

Upgrade Notes:

+ +

Developer Notes:

+ +

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

Commit Details:

+

Initial commit for certified-22.8

+

Author: George Joseph + Date: 2026-02-23

+

xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.

+

Author: George Joseph + Date: 2026-01-15

+

The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external + entities, could allow a potential XXE injection attack. Replacing it with + XML_PARSE_NONET, which prevents network access, is safer.

+

Resolves: #GHSA-85x7-54wr-vh42

+

ast_coredumper: check ast_debug_tools.conf permissions

+

Author: Mike Bradeen + Date: 2026-01-15

+

Prevent ast_coredumper from using ast_debug_tools.conf files that are + not owned by root or are writable by other users or groups.

+

Prevent ast_logescalator and ast_loggrabber from doing the same if + they are run as root.

+

Resolves: #GHSA-rvch-3jmx-3jf3

+

UserNote: ast_debug_tools.conf must be owned by root and not be + writable by other users or groups to be used by ast_coredumper or + by ast_logescalator or ast_loggrabber when run as root.

+

http.c: Change httpstatus to default disabled and sanitize output.

+

Author: George Joseph + Date: 2026-01-15

+

To address potential security issues, the httpstatus page is now disabled + by default and the echoed query string and cookie output is html-escaped.

+

Resolves: #GHSA-v6hp-wh3r-cwxh

+

UpgradeNote: To prevent possible security issues, the /httpstatus page + served by the internal web server is now disabled by default. To explicitly + enable it, set enable_status=yes in http.conf.

+

ast_coredumper: create gdbinit file with restrictive permissions

+

Author: Mike Bradeen + Date: 2026-01-15

+

Modify gdbinit to use the install command with explicit permissions (-m 600) + when creating the .ast_coredumper.gdbinit file. This ensures the file is + created with restricted permissions (readable/writable only by the owner) + to avoid potential privilege escalation.

+

Resolves: #GHSA-xpc6-x892-v83c

+

asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.

+

Author: Sean Bright + Date: 2026-01-23

+

Resolves: #1739

+

chan_websocket.conf.sample: Fix category name.

+

Author: George Joseph + Date: 2026-01-21

+

UserNote: The category name in the chan_websocket.conf.sample file was + incorrect. It should be "global" instead of "general".

+

chan_websocket: Fixed Ping/Pong messages hanging up the websocket channel

+

Author: Joe Garlick + Date: 2026-01-15

+

When chan_websocket received a Ping or a Pong opcode it would cause the channel to hangup. This change allows Ping/Pong opcodes and allows them to silently pass

+

cli.c: Allow 'channel request hangup' to accept patterns.

+

Author: Sean Bright + Date: 2026-01-05

+

This extends 'channel request hangup' to accept multiple channel + names, a POSIX Extended Regular Expression, a glob-like pattern, or a + combination of all of them.

+

UserNote: The 'channel request hangup' CLI command now accepts + multiple channel names, POSIX Extended Regular Expressions, glob-like + patterns, or a combination of all of them. See the CLI command 'core + show help channel request hangup' for full details.

+

res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache populate command

+

Author: Mike Bradeen + Date: 2026-01-06

+

Reduce cache lock time for AMI and CLI sorcery memory cache populate + commands by adding a new populate_lock to the sorcery_memory_cache + struct which is locked separately from the existing cache lock so that + the cache lock can be maintained for a reduced time, locking only when + the cache objects are removed and re-populated.

+

Resolves: #1700

+

UserNote: The AMI command sorcery memory cache populate will now + return an error if there is an internal error performing the populate. + The CLI command will display an error in this case as well.

+

Add comment to asterisk.conf.sample clarifying that template sections are ignored

+

Author: phoneben + Date: 2026-01-05

+

Add comment to asterisk.conf.sample clarifying that template sections are ignored.

+

Resolves: #1692

+

chan_websocket: Use the channel's ability to poll fds for the websocket read.

+

Author: George Joseph + Date: 2025-12-30

+

We now add the websocket's file descriptor to the channel's fd array and let + it poll for data availability instead if having a dedicated thread that + does the polling. This eliminates the thread and allows removal of most + explicit locking since the core channel code will lock the channel to prevent + simultaneous calls to webchan_read, webchan_hangup, etc.

+

While we were here, the hangup code was refactored to use ast_hangup_with_cause + instead of directly queueing an AST_CONTROL_HANGUP frame. This allows us + to set hangup causes and generate snapshots.

+

For a bit of extra debugging, a table of websocket close codes was added + to http_websocket.h with an accompanying "to string" function added to + res_http_websocket.c

+

Resolves: #1683

+

asterisk.c: Allow multi-byte characters on the Asterisk CLI.

+

Author: Sean Bright + Date: 2025-12-13

+

Versions of libedit that support Unicode expect that the + EL_GETCFN (the function that does character I/O) will fill in a + wchar_t with a character, which may be multi-byte. The built-in + function that libedit provides, but does not expose with a public API, + does properly handle multi-byte sequences.

+

Due to the design of Asterisk's console processing loop, Asterisk + provides its own implementation which does not handle multi-byte + characters. Changing Asterisk to use libedit's built-in function would + be ideal, but would also require changing some fundamental things + about console processing which could be fairly disruptive.

+

Instead, we bring in libedit's read_char implementation and modify + it to suit our specific needs.

+

Resolves: #60

+

func_presencestate.c: Allow NOT_SET to be set from CLI.

+

Author: Sean Bright + Date: 2026-01-01

+

Resolves: #1647

+

res/ari/resource_bridges.c: Normalize channel_format ref handling for bridge media

+

Author: Peter Krall + Date: 2025-12-17

+

Always take an explicit reference on the format used for bridge playback + and recording channels, regardless of where it was sourced, and release + it after prepare_bridge_media_channel. This aligns the code paths and + avoids mixing borrowed and owned references while preserving behavior.

+

Fixes: #1648

+

res_geolocation: Fix multiple issues with XML generation.

+

Author: George Joseph + Date: 2025-12-17

+ +

In addition to fixing the above, several other code refactorings were + performed and the unit test enhanced to include a round trip + XML -> eprofile -> XML validation.

+

Resolves: #1667

+

UserNote: Geolocation: Two new optional profile parameters have been added. + * pidf_element_id which sets the value of the id attribute on the top-level + PIDF-LO device, person or tuple elements. + * device_id which sets the content of the <deviceID> element. + Both parameters can include channel variables.

+

UpgradeNote: Geolocation: In order to correct bugs in both code and + documentation, the following changes to the parameters for GML geolocation + locations are now in effect: + * The documented but unimplemented crs (coordinate reference system) element + has been added to the location_info parameter that indicates whether the 2d + or 3d reference system is to be used. If the crs isn't valid for the shape + specified, an error will be generated. The default depends on the shape + specified. + * The Circle, Ellipse and ArcBand shapes MUST use a 2d crs. If crs isn't + specified, it will default to 2d for these shapes. + The Sphere, Ellipsoid and Prism shapes MUST use a 3d crs. If crs isn't + specified, it will default to 3d for these shapes. + The Point and Polygon shapes may use either crs. The default crs is 2d + however so if 3d positions are used, the crs must be explicitly set to 3d. + * The geoloc show gml_shape_defs CLI command has been updated to show which + coordinate reference systems are valid for each shape. + * The pos3d element has been removed in favor of allowing the pos element + to include altitude if the crs is 3d. The number of values in the pos + element MUST be 2 if the crs is 2d and 3 if the crs is 3d. An error + will be generated for any other combination. + * The angle unit-of-measure for shapes that use angles should now be included + in the respective parameter. The default is degrees. There were some + inconsistent references to orientation_uom in some documentation but that + parameter never worked and is now removed. See examples below. + Examples... + location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20" + location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620" + location_info = shape="Point", pos="39.0 -105.0" + location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20" + semiMinorAxis="10", verticalAxis="0", orientation="25 degrees" + pidf_element_id = ${CHANNEL(name)}-${EXTEN} + device_id = mac:001122334455 + Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})

+

stasis/control.c: Add destructor to timeout_datastore.

+

Author: George Joseph + Date: 2025-12-31

+

The timeout_datastore was missing a destructor resulting in a leak + of 16 bytes for every outgoing ARI call.

+

Resolves: #1681

+

func_talkdetect.c: Remove reference to non-existent variables.

+

Author: Sean Bright + Date: 2025-12-30

+

configure.ac: use AC_PATH_TOOL for nm

+

Author: Nathaniel Wesley Filardo + Date: 2025-11-27

+

nm might, especially in cross-compilation scenarios, be available but prefixed with the target triple. So: use AC_PATH_TOOL rather than AC_PATH_PROG to find it. (See https://www.gnu.org/software/autoconf/manual/autoconf-2.68/html_node/Generic-Programs.html .)

+

Found and proposed fix tested by cross-compiling Asterisk using Nixpkgs on x86_64 targeting aarch64. :)

+

res_pjsip_mwi: Fix off-nominal endpoint ao2 ref leak in mwi_get_notify_data

+

Author: Alexei Gradinari + Date: 2025-12-29

+

Delay acquisition of the ast_sip_endpoint reference in mwi_get_notify_data() + to avoid an ao2 ref leak on early-return error paths.

+

Move ast_sip_subscription_get_endpoint() to just before first use so all + acquired references are properly cleaned up.

+

Fixes: #1675

+

res_pjsip_messaging: Add support for following 3xx redirects

+

Author: Maximilian Fridrich + Date: 2025-11-07

+

This commit integrates the redirect module into res_pjsip_messaging + to enable following 3xx redirect responses for outgoing SIP MESSAGEs.

+

When follow_redirect_methods contains 'message' on an endpoint, Asterisk + will now follow 3xx redirect responses for MESSAGEs, similar to how + it behaves for INVITE responses.

+

Resolves: #1576

+

UserNote: A new pjsip endpoint option follow_redirect_methods was added. + This option is a comma-delimited, case-insensitive list of SIP methods + for which SIP 3XX redirect responses are followed. An alembic upgrade + script has been added for adding this new option to the Asterisk + database.

+

res_pjsip: Introduce redirect module for handling 3xx responses

+

Author: Maximilian Fridrich + Date: 2025-11-07

+

This commit introduces a new redirect handling module that provides + infrastructure for following SIP 3xx redirect responses. The redirect + functionality respects the endpoint's redirect_method setting and only + follows redirects when set to 'uri_pjsip'. This infrastructure can be + used by any PJSIP module that needs to handle 3xx redirect responses.

+

app_mixmonitor.c: Fix crash in mixmonitor_ds_remove_and_free when datastore is NULL

+

Author: Tinet-mucw + Date: 2025-12-25

+

The datastore may be NULL, so a null pointer check needs to be added.

+

Resolves: #1673

+

res_pjsip_refer: don't defer session termination for ari transfer

+

Author: Sven Kube + Date: 2025-10-23

+

Allow session termination during an in progress ari handled transfer.

+

chan_dahdi.conf.sample: Avoid warnings with default configs.

+

Author: Naveen Albert + Date: 2025-10-23

+

callgroup and pickupgroup may only be specified for FXO-signaled channels; + however, the chan_dahdi sample config had these options uncommented in + the [channels] section, thus applying these settings to all channels, + resulting in warnings. Comment these out so there are no warnings with + an unmodified sample config.

+

Resolves: #1552

+

main/dial.c: Set channel hangup cause on timeout in handle_timeout_trip

+

Author: sarangr7 + Date: 2025-12-18

+

When dial attempts timeout in the core dialing API, the channel's hangup + cause was not being set before hanging up. Only the ast_dial_channel + structure's internal cause field was updated, but the actual ast_channel + hangup cause remained unset.

+

This resulted in incorrect or missing hangup cause information being + reported through CDRs, AMI events, and other mechanisms that read the + channel's hangup cause when dial timeouts occurred via applications + using the dialing API (FollowMe, Page, etc.).

+

The fix adds proper channel locking and sets AST_CAUSE_NO_ANSWER on + the channel before calling ast_hangup(), ensuring consistent hangup + cause reporting across all interfaces.

+

Resolves: #1660

+

cel: Add missing manager documentation.

+

Author: Sean Bright + Date: 2025-12-12

+

The LOCAL_OPTIMIZE_BEGIN, STREAM_BEGIN, STREAM_END, and DTMF CEL + events were not all documented in the CEL configuration file or the + manager documentation for the CEL event.

+

res_odbc: Use SQL_SUCCEEDED() macro where applicable.

+

Author: Sean Bright + Date: 2025-12-17

+

This is just a cleanup of some repetitive code.

+

rtp/rtcp: Configure dual-stack behavior via IPV6_V6ONLY

+

Author: Justin T. Gibbs + Date: 2025-12-21

+

Dual-stack behavior (simultaneous listening for IPV4 and IPV6 + connections on a single socket) is required by Asterisk's ICE + implementation. On systems with the IPV6_V6ONLY sockopt, set + the option to 0 (dual-stack enabled) when binding to the IPV6 + any address. This ensures correct behavior regardless of the + system's default dual-stack configuration.

+

http.c: Include remote address in URI handler message.

+

Author: Sean Bright + Date: 2025-12-22

+

Resolves: #1662

+

Disable device state caching for ephemeral channels

+

Author: phoneben + Date: 2025-12-09

+

chan_audiosocket/chan_rtp/res_stasis_snoop: Disable device state caching for ephemeral channels

+

Resolves: #1638

+

chan_websocket: Add locking in send_event and check for NULL websocket handle.

+

Author: George Joseph + Date: 2025-12-10

+

On an outbound websocket connection, when the triggering caller hangs up, + webchan_hangup() closes the outbound websocket session and sets the websocket + session handle to NULL. If the hangup happened in the tiny window between + opening the outbound websocket connection and before read_thread_handler() + was able to send the MEDIA_START message, it could segfault because the + websocket session handle was NULL. If it didn't actually segfault, there was + also the possibility that the websocket instance wouldn't get cleaned up which + could also cause the channel snapshot to not get cleaned up. That could + cause memory leaks and core show channels to list phantom WebSocket + channels.

+

To prevent the race, the send_event() macro now locks the websocket_pvt + instance and checks the websocket session handle before attempting to send + the MEDIA_START message.

+

Resolves: #1643 + Resolves: #1645

+

Fix false null-deref warning in channel_state

+

Author: phoneben + Date: 2025-12-08

+

Resolve analyzer warning in channel_state by checking AST_FLAG_DEAD on snapshot, which is guaranteed non-NULL.

+

Resolves: #1430

+

endpoint.c: Plug a memory leak in ast_endpoint_shutdown().

+

Author: George Joseph + Date: 2025-12-08

+

Commit 26795be introduced a memory leak of ast_endpoint when + ast_endpoint_shutdown() was called. The leak occurs only if a configuration + change removes an endpoint and isn't related to call volume or the length of + time asterisk has been running. An ao2_ref(-1) has been added to + ast_endpoint_shutdown() to plug the leak.

+

Resolves: #1635

+

Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()"

+

Author: Sean Bright + Date: 2025-12-03

+

This reverts commit 517766299093d7a9798af68b39951ed8b2469836.

+

For rationale, see #1621 and #1606

+

cel_manager.c: Correct manager event mask for CEL events.

+

Author: Sean Bright + Date: 2025-12-05

+

There is no EVENT_FLAG_CEL and these events are raised with as + EVENT_FLAG_CALL.

+

app_queue.c: Update docs to correct QueueMemberPause event name.

+

Author: Sean Bright + Date: 2025-12-04

+

taskprocessors: Improve logging and add new cli options

+

Author: Mike Bradeen + Date: 2025-10-28

+

This change makes some small changes to improve log readability in + addition to the following changes:

+

Modified 'core show taskprocessors' to now show Low time and High time + for task execution.

+

New command 'core show taskprocessor name ' to dump + taskprocessor info and current queue.

+

Addionally, a new test was added to demonstrate the 'show taskprocessor + name' functionality: + test execute category /main/taskprocessor/ name taskprocessor_cli_show

+

Setting 'core set debug 3 taskprocessor.c' will now log pushed tasks. + (Warning this is will cause extremely high levels of logging at even + low traffic levels.)

+

Resolves: #1566

+

UserNote: New CLI command has been added - + core show taskprocessor name

+

manager: fix double free of criteria variable when adding filter

+

Author: Michal Hajek + Date: 2025-10-13

+

Signed-off-by: Michal Hajek michal.hajek@daktela.com

+

Fixes: #1531

+

app_stream_echo.c: Check that stream is non-NULL before dereferencing.

+

Author: Sean Bright + Date: 2025-12-01

+

Also re-order and rename the arguments of stream_echo_write_error to + match those of ast_write_stream for consistency.

+

Resolves: #1427

+

abstract_jb.c: Remove redundant timer check per static analysis.

+

Author: Sean Bright + Date: 2025-12-01

+

While this check is technically unnecessary, it also was not harmful.

+

The 2 other items mentioned in the linked issue are false positives + and require no action.

+

Resolves: #1417

+

channelstorage_cpp: Fix fallback return value in channelstorage callback

+

Author: phoneben + Date: 2025-11-26

+

callback returned the last iterated channel when no match existed, causing invalid channel references and potential double frees. Updated to correctly return NULL when there is no match.

+

Resolves: #1609

+

ccss: Add option to ccss.conf to globally disable it.

+

Author: George Joseph + Date: 2025-11-19

+

The Call Completion Supplementary Service feature is rarely used but many of + it's functions are called by app_dial and channel.c "just in case". These + functions lock and unlock the channel just to see if CCSS is enabled on it, + which it isn't 99.99% of the time.

+

UserNote: A new "enabled" parameter has been added to ccss.conf. It defaults + to "yes" to preserve backwards compatibility but CCSS is rarely used so + setting "enabled = no" in the "general" section can save some unneeded channel + locking operations and log message spam. Disabling ccss will also prevent + the func_callcompletion and chan_dahdi modules from loading.

+

DeveloperNote: A new API ast_is_cc_enabled() has been added. It should be + used to ensure that CCSS is enabled before making any other ast_cc_* calls.

+

app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.

+

Author: George Joseph + Date: 2025-11-20

+

UpgradeNote: In an effort to reduce log spam, two normal progress + "pickup attempted" log messages from app_directed_pickup have been changed + from NOTICE to VERBOSE(3). This puts them on par with other normal + dialplan progress messages.

+

chan_websocket: Fix crash on DTMF_END event.

+

Author: Sean Bright + Date: 2025-11-20

+

Resolves: #1604

+

chan_websocket.c: Tolerate other frame types

+

Author: Joe Garlick + Date: 2025-11-12

+

Currently, if chan_websocket receives an un supported frame like comfort noise it will exit the websocket. The proposed change is to tolerate the other frames by not sending them down the websocket but instead just ignoring them.

+

Resolves: #1587

+

app_reload: Fix Reload() without arguments.

+

Author: Naveen Albert + Date: 2025-11-17

+

Calling Reload() without any arguments is supposed to reload + everything (equivalent to a 'core reload'), but actually does + nothing. This is because it was calling ast_module_reload with + an empty string, and the argument needs to explicitly be NULL.

+

Resolves: #1597

+

pbx.c: Print new context count when reloading dialplan.

+

Author: Naveen Albert + Date: 2025-11-17

+

When running "dialplan reload", the number of contexts reported + is initially wrong, as it is the old context count. Running + "dialplan reload" a second time returns the correct number of + contexts that are loaded. This can confuse users into thinking + that the reload didn't work successfully the first time.

+

This counter is currently only incremented when iterating the + old contexts prior to the context merge; at the very end, get + the current number of elements in the context hash table and + report that instead. This way, the count is correct immediately + whenever a reload occurs.

+

Resolves: #1599

+

Makefile: Add module-list-* targets.

+

Author: C. Maj + Date: 2025-11-17

+

Convenience wrappers for showing modules at various support levels.

+ +

Resolves: #1572

+

UserNote: Try "make module-list-deprecated" to see what modules + are on their way out the door.

+

app_disa: Avoid use of removed ResetCDR() option.

+

Author: Naveen Albert + Date: 2025-11-14

+

Commit a46d5f9b760f84b9f27f594b62507c1443aa661b removed the deprecated + 'e' option to ResetCDR; this now causes DISA() to emit a warning + if attempting to call ResetCDR() with the deprecated option (in + all cases except when the no answer option is provided). Rewrite + the code to do this the current way.

+

Resolves: #1592

+

core_unreal.c: Use ast instead of p->chan to get the DIALSTATUS variable

+

Author: Tinet-mucw + Date: 2025-11-13

+

After p->chan = NULL, ast still points to the valid channel object, + using ast safely accesses the channel's DIALSTATUS variable before it's fully destroyed

+

Resolves: #1590

+

ast_coredumper: Fix multiple issues

+

Author: George Joseph + Date: 2025-11-07

+ +

In the embedded gdbinit script:

+ +

app_mixmonitor: Add 's' (skip) option to delay recording.

+

Author: Daouda Taha + Date: 2025-10-28

+

The 's' (skip) option delays MixMonitor recording until the specified number of seconds + (can be fractional) have elapsed since MixMonitor was invoked.

+

No audio is written to the recording file during this time. If the call ends before this + period, no audio will be saved. This is useful for avoiding early audio such as + announcements, ringback tones, or other non-essential sounds.

+

UserNote: This change introduces a new 's()' (skip) option to the MixMonitor + application. Example: + MixMonitor(${UNIQUEID}.wav,s(3))

+

This skips recording for the first 3 seconds before writing audio to the file. + Existing MixMonitor behavior remains unchanged when the 's' option is not used.

+

stasis: switch stasis show topics temporary container from list - RBtree

+

Author: phoneben + Date: 2025-11-11

+

switch stasis show topics temporary container from list to RB-tree + minimizing lock time

+

Resolves: #1585

+

app_dtmfstore: Avoid a potential buffer overflow.

+

Author: Sean Bright + Date: 2025-11-07

+

Prefer snprintf() so we can readily detect if our output was + truncated.

+

Resolves: #1421

+

main: Explicitly mark case statement fallthrough as such.

+

Author: Sean Bright + Date: 2025-11-07

+

Resolves: #1442

+

bridge_softmix: Return early on topology allocation failure.

+

Author: Sean Bright + Date: 2025-11-07

+

Resolves: #1446

+

bridge_simple: Increase code verbosity for clarity.

+

Author: Sean Bright + Date: 2025-11-07

+

There's no actual problem here, but I can see how it might by + confusing.

+

Resolves: #1444

+

app_queue.c: Only announce to head caller if announce_to_first_user

+

Author: Kristian F. Høgh + Date: 2025-10-30

+

Only make announcements to head caller if announce_to_first_user is true

+

Fixes: #1568

+

UserNote: When announce_to_first_user is false, no announcements are played to the head caller

+

chan_websocket: Add ability to place a MARK in the media stream.

+

Author: George Joseph + Date: 2025-11-05

+

Also cleaned up a few unused #if blocks, and started sending a few ERROR + events back to the apps.

+

Resolves: #1574

+

DeveloperNote: Apps can now send a MARK_MEDIA command with an optional + correlation_id parameter to chan_websocket which will be placed in the + media frame queue. When that frame is dequeued after all intervening media + has been played to the core, chan_websocket will send a + MEDIA_MARK_PROCESSED event to the app with the same correlation_id + (if any).

+

chan_websocket: Add capability for JSON control messages and events.

+

Author: George Joseph + Date: 2025-10-22

+

With recent enhancements to chan_websocket, the original plain-text + implementation of control messages and events is now too limiting. We + probably should have used JSON initially but better late than never. Going + forward, enhancements that require control message or event changes will + only be done to the JSON variants and the plain-text variants are now + deprecated but not yet removed.

+ +

The JSON for commands sent by the app to Asterisk must be... + { "command": "<command>" ... } where <command> is one of ANSWER, HANGUP, + START_MEDIA_BUFFERING, etc. The STOP_MEDIA_BUFFERING command takes an + additional, optional parameter to be returned in the corresponding + MEDIA_BUFFERING_COMPLETED event: + { "command": "STOP_MEDIA_BUFFERING", "correlation_id": "<correlation id>" }.

+

The JSON for events sent from Asterisk to the app will be... + { "event": "<event>", "channel_id": "<channel_id>" ... }. + The MEDIA_START event will now look like...

+

{ + "event": "MEDIA_START", + "connection_id": "media_connection1", + "channel": "WebSocket/media_connection1/0x5140001a0040", + "channel_id": "1761245643.1", + "format": "ulaw", + "optimal_frame_size": 160, + "ptime": 20, + "channel_variables": { + "DIALEDPEERNUMBER": "media_connection1/c(ulaw)", + "MEDIA_WEBSOCKET_CONNECTION_ID": "media_connection1", + "MEDIA_WEBSOCKET_OPTIMAL_FRAME_SIZE": "160" + } + }

+

Note the addition of the channel variables which can't be supported + with the plain-text formatting.

+

The documentation will be updated with the exact formats for all commands + and events.

+

Resolves: #1546 + Resolves: #1563

+

DeveloperNote: The chan_websocket plain-text control and event messages are now + deprecated (but remain the default) in favor of JSON formatted messages. + See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for + more information.

+

DeveloperNote: A "transport_data" parameter has been added to the + channels/externalMedia ARI endpoint which, for websocket, allows the caller + to specify parameters to be added to the dialstring for the channel. For + instance, "transport_data": "f(json)".

+

build: Add menuselect options to facilitate code tracing and coverage

+

Author: George Joseph + Date: 2025-10-30

+

The following options have been added to the menuselect "Compiler Flags" + section...

+

CODE_COVERAGE: The ability to enable code coverage via the --enable-coverage + configure flag has existed for many years but changing it requires + re-running ./configure which is painfully slow. With this commit, you can + now enable and disable it via menuselect. Setting this option adds the + -ftest-coverage and -fprofile-arcs flags on the gcc and ld command lines. + It also sets DONT_OPTIMIZE. Note: If you use the --enable-coverage configure + flag, you can't turn it off via menuselect so choose one method and stick to + it.

+

KEEP_FRAME_POINTERS: This option sets -fno-omit-frame-pointers on the gcc + command line which can facilitate debugging with 'gdb' and tracing with 'perf'. + Unlike CODE_COVERAGE, this option doesn't depend on optimization being + disabled. It does however conflict with COMPILE_DOUBLE.

+

channelstorage: Allow storage driver read locking to be skipped.

+

Author: George Joseph + Date: 2025-11-06

+

After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI + channels/externalMedia was called with a custom channel id AND the + cpp_map_name_id channel storage backend is in use, a deadlock can occur when + hanging up the channel. It's actually triggered in + channel.c:__ast_channel_alloc_ap() when it gets a write lock on the + channelstorage driver then subsequently does a lookup for channel uniqueid + which now does a read lock. This is an invalid operation and causes the lock + state to get "bad". When the channels try to hang up, a write lock is + attempted again which hangs and causes the deadlock.

+

Now instead of the cpp_map_name_id channelstorage driver "get" APIs + automatically performing a read lock, they take a "lock" parameter which + allows a caller who already has a write lock to indicate that the "get" API + must not attempt its own lock. This prevents the state from getting mesed up.

+

The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't + have this issue but since it also implements the common channelstorage API, + it needed its "get" implementations updated to take the lock parameter. They + just don't use it.

+

Resolves: #1578

+

res_audiosocket: fix temporarily unavailable

+

Author: Roman Pertsev + Date: 2025-10-07

+

Operations on non-blocking sockets may return a resource temporarily unavailable error (EAGAIN or EWOULDBLOCK). This is not a fatal error but a normal condition indicating that the operation would block.

+

This patch corrects the handling of this case. Instead of incorrectly treating it as a reason to terminate the connection, the code now waits for data to arrive on the socket.

+

safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.

+

Author: Sean Bright + Date: 2025-10-22

+ +

Resolves: #1554

+

res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.

+

Author: George Joseph + Date: 2025-10-24

+

Also...

+ +

Resolves: #781

+

UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added + which will allow suppressing attestation on a call-by-call basis + regardless of the profile attached to the outgoing endpoint.

+

iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.

+

Author: Tinet-mucw + Date: 2025-10-26

+

The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). + In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake.

+

Resolves: #1559

+

chan_pjsip: Disable SSRC change for WebRTC endpoints.

+

Author: George Joseph + Date: 2025-10-21

+

Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with + blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC + clients that are sensitive to SSRC changes and non-monotonic timestamps so + the fix is now disabled for endpoints with the "bundle" parameter set to true.

+

Resolves: #1535

+

chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.

+

Author: gauravs456 + Date: 2025-10-21

+

Resolves: #1544

+

safe_asterisk: Fix logging and sorting issue.

+

Author: George Joseph + Date: 2025-10-17

+

Re-enabled "TTY=9" which was erroneously disabled as part of a recent + security fix and removed another logging "fix" that was added.

+

Also added a sort to the "find" that enumerates the scripts to be sourced so + they're sourced in the correct order.

+

Resolves: #1539

+

Fix Endianness detection in utils.h for non-Linux

+

Author: Christoph Moench-Tegeder + Date: 2025-10-19

+

Commit 43bf8a4ded7a65203b766b91eaf8331a600e9d8d introduced endian + dependend byte-swapping code in include/asterisk/utils.h, where the + endianness was detected using the __BYTE_ORDER macro. This macro + lives in endian.h, which on Linux is included implicitely (by the + network-related headers, I think), but on FreeBSD the headers are + laid out differently and we do not get __BYTE_ORDER the implicit way.

+

Instead, this makes the usage of endian.h explicit by including it + where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the + POSIX-defined ones (see + https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html + for standard compliance). Additionally, this adds a compile-time check + for the endianness-logic: compilation will fail if neither big nor + little endian can be detected.

+

Fixes: #1536

+

app_queue.c: Fix error in Queue parameter documentation.

+

Author: Ben Ford + Date: 2025-10-20

+

When macro was removed in Asterisk 21, the parameter documentation in + code was not updated to reflect the correct numerization for gosub. It + still stated that it was the seventh parameter, but got shifted to the + sixth due to the removal of macro. This has been updated to correctly + reflect the parameter order, and a note has been added to the XML that + states this was done after the initial commit.

+

Fixes: #1534

+

UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk. + This resulted in argument order changing for the Queue dialplan + application since the macro argument was removed. Upgrade notice was + missed when this was done, so this upgrade note has been added to + provide a record of such and a notice to users who may have not upgraded + yet.

+

devicestate: Don't publish redundant device state messages.

+

Author: Joshua C. Colp + Date: 2025-10-17

+

When publishing device state check the local cache for the + existing device state. If the new device state is unchanged + from the prior one, don't bother publishing the update. This + can reduce the work done by consumers of device state, such + as hints and app_queue, by not publishing a message to them.

+

These messages would most often occur with devices that are + seeing numerous simultaneous channels. The underlying device + state would remain as in use throughout, but an update would + be published as channels are created and hung up.

+

chan_pjsip: Add technology-specific off-nominal hangup cause to events.

+

Author: George Joseph + Date: 2025-10-14

+

Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI + and AMI hangup and channel destroyed events, it can be helpful to know what + the actual channel technology code was if the call was unsuccessful. + For PJSIP, it's the SIP response code.

+ +

DeveloperNote: A "tech_cause" parameter has been added to the + ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" + parameter has been added to the HangupRequest, SoftHangupRequest and Hangup + AMI event messages. For chan_pjsip, these will be set to the last SIP + response status code for off-nominally terminated calls. The parameter is + suppressed for nominal termination.

+

res_audiosocket: add message types for all slin sample rates

+

Author: Sven Kube + Date: 2025-10-10

+

Extend audiosocket messages with types 0x11 - 0x18 to create asterisk + frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 format, enabling the transmission of audio at a higher sample + rates. For audiosocket messages sent by Asterisk, the message kind is + determined by the format of the originating asterisk frame.

+

UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added + for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 audio. External applications using audiosocket may need to be + updated to support these message types if the audiosocket channel is + created with one of these audio formats.

+

res_fax.c: lower FAXOPT read warning to debug level

+

Author: phoneben + Date: 2025-10-03

+

Reading ${FAXOPT()} before a fax session is common in dialplans to check fax state. + Currently this logs an error even when no fax datastore exists, creating excessive noise. + Change these messages to ast_debug(3, …) so they appear only with debug enabled.

+

Resolves: #1509

+

endpoints: Remove need for stasis subscription.

+

Author: Joshua C. Colp + Date: 2025-10-10

+

When an endpoint is created in the core of Asterisk a subscription + was previously created alongside it to monitor any channels being + destroyed that were related to it. This was done by receiving all + channel snapshot updates for every channel and only reacting when + it was indicated that the channel was dead.

+

This change removes this logic and instead provides an API call + for directly removing a channel from an endpoint. This is called + when channels are destroyed. This operation is fast, so blocking + the calling thread for a short period of time doesn't have any + noticeable impact.

+

app_queue: Allow stasis message filtering to work.

+

Author: Joshua C. Colp + Date: 2025-10-10

+

The app_queue module subscribes on a per-dialed agent basis to both + the bridge all and channel all topics to keep apprised of things going + on involving them. This subscription has associated state that must + be cleaned up when the subscription ends. This was done by setting + a default router callback that only had logic to handle the case + where the subscription ends. By using the default router callback + all filtering for the subscription was disabled, causing unrelated + messages to get published and handled by it.

+

This change makes it so that an explicit route is added for the + message type used for the message indicating the subscription has + ended and removes the default router callback. This allows message + filtering to occur on publishing reducing the messages to app_queue + to only those it is interested in.

+

taskpool: Fix some references to threadpool that should be taskpool.

+

Author: George Joseph + Date: 2025-10-10

+

Resolves: #1478

+

Update contact information for anthm

+

Author: Anthony Minessale + Date: 2025-10-10

+

chan_websocket.c: Change payload references to command instead.

+

Author: George Joseph + Date: 2025-10-08

+

Some of the tests in process_text_message() were still comparing to the + websocket message payload instead of the "command" string.

+

Resolves: #1525

+

func_callerid: Document limitation of DNID fields.

+

Author: Naveen Albert + Date: 2025-10-06

+

The Dial() application does not propagate DNID fields, which is counter + to the behavior of the other Caller ID fields. This behavior is likely + intentional since the use of Dial theoretically suggests a new dialed + number, but document this caveat to inform users of it.

+

Resolves: #1519

+

func_channel: Allow R/W of ADSI CPE capability setting.

+

Author: Naveen Albert + Date: 2025-10-06

+

Allow retrieving and setting the channel's ADSI capability from the + dialplan.

+

Resolves: #1514

+

UserNote: CHANNEL(adsicpe) can now be read or written to change + the channels' ADSI CPE capability setting.

+

core_unreal: Preserve ADSI capability when dialing Local channels.

+

Author: Naveen Albert + Date: 2025-10-06

+

Dial() already preserves the ADSI capability by copying it to the new + channel, but since Local channel pairs consist of two channels, we + also need to copy the capability to the second channel.

+

Resolves: #1517

+

func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()

+

Author: Igor Goncharovsky + Date: 2025-09-04

+

As soon as SIP call may end with several Reason headers, we + want to make all of them available through the HAGUPCAUSE() function. + This implementation uses the same ao2 hash for cause codes storage + and adds a flag to make difference between last processed sip + message and content of reason headers.

+

UserNote: Added a new option to HANGUPCAUSE to access additional + information about hangup reason. Reason headers from pjsip + could be read using 'tech_extended' cause type.

+

sig_analog: Allow '#' to end the inter-digit timeout when dialing.

+

Author: Naveen Albert + Date: 2025-10-03

+

It is customary to allow # to terminate digit collection immediately + when there would normally be a timeout. However, currently, users are + forced to wait for the timeout to expire when dialing numbers that + are prefixes of other valid matches, and there is no way to end the + timeout early. Customarily, # terminates the timeout, but at the moment, + this is just rejected unless there happens to be a matching extension + ending in #.

+

Allow # to terminate the timeout in cases where there is no dialplan + match. This ensures that the dialplan is always respected, but if a + valid extension has been dialed that happens to prefix other valid + matches, # can be used to dial it immediately.

+

Resolves: #1510

+

func_math: Add DIGIT_SUM function.

+

Author: Naveen Albert + Date: 2025-10-01

+

Add a function (DIGIT_SUM) which returns the digit sum of a number.

+

Resolves: #1499

+

UserNote: The DIGIT_SUM function can be used to return the digit sum of + a number.

+

app_sf: Add post-digit timer option to ReceiveSF.

+

Author: Naveen Albert + Date: 2025-10-01

+

Add a sorely needed option to set a timeout between digits, rather than + for receiving the entire number. This is needed if the number of digits + being sent is unknown by the receiver in advance. Previously, we had + to wait for the entire timer to expire.

+

Resolves: #1493

+

UserNote: The 't' option for ReceiveSF now allows for a timer since + the last digit received, in addition to the number-wide timeout.

+

codec_builtin.c: Adjust some of the quality scores to reflect reality.

+

Author: Naveen Albert + Date: 2025-10-02

+

Among the lower-quality voice codecs, some of the quality scores did + not make sense relative to each other.

+

For instance, quality-wise, G.729 > G.723 > PLC10. + However, current scores do not uphold these relationships.

+

Tweak the scores slightly to reflect more accurate relationships.

+

Resolves: #1501

+

res_tonedetect: Fix formatting of XML documentation.

+

Author: Naveen Albert + Date: 2025-10-02

+

Fix the indentation in the documentation for the variable list.

+

Resolves: #1507

+

res_fax: Add XML documentation for channel variables.

+

Author: Naveen Albert + Date: 2025-10-02

+

Document the channel variables currently set by SendFAX and ReceiveFAX.

+

Resolves: #1505

+

channelstorage_cpp_map_name_id: Add read locking around retrievals.

+

Author: George Joseph + Date: 2025-10-01

+

When we retrieve a channel from a C++ map, we actually get back a wrapper + object that points to the channel then right after we retrieve it, we bump its + reference count. There's a tiny chance however that between those two + statements a delete and/or unref might happen which would cause the wrapper + object or the channel itself to become invalid resulting in a SEGV. To avoid + this we now perform a read lock on the driver around those statements.

+

Resolves: #1491

+

app_dial: Allow fractional seconds for dial timeouts.

+

Author: Naveen Albert + Date: 2025-09-30

+

Even though Dial() internally uses milliseconds for its dial timeouts, + this capability has been mostly obscured from users as the argument is + only parsed as an integer, thus forcing the use of whole seconds for + timeouts.

+

Parse it as a decimal instead so that timeouts can now truly have + millisecond precision.

+

Resolves: #1487

+

UserNote: The answer and progress dial timeouts now have millisecond + precision, instead of having to be whole numbers.

+

dsp.c: Make minor fixes to debug log messages.

+

Author: Naveen Albert + Date: 2025-10-01

+

Commit dc8e3eeaaf094a3d16991289934093d5e7127680 improved the debug log + messages in dsp.c. This makes two minor corrections to it:

+ +

Resolves: #1496

+

config_options.c: Improve misleading warning.

+

Author: Naveen Albert + Date: 2025-09-30

+

When running "config show help ", if no XML documentation exists + for the specified module, "Module not found." is returned, + which is misleading if the module is loaded but simply has no XML + documentation for its config. Improve the message to clarify that the + module may simply have no config documentation.

+

Resolves: #1489

+

func_scramble: Add example to XML documentation.

+

Author: Naveen Albert + Date: 2025-09-29

+

The previous lack of an example made it ambiguous if the arguments went + inside the function arguments or were part of the right-hand value.

+

Resolves: #1485

+

sig_analog: Eliminate potential timeout with Last Number Redial.

+

Author: Naveen Albert + Date: 2025-09-29

+

If Last Number Redial is used to redial, ensure that we do not wait + for further digits. This was possible if the number that was last + dialed is a prefix of another possible dialplan match. Since all we + did is copy the number into the extension buffer, if other matches + are now possible, there would thus be a timeout before the call went + through. We now complete redialed calls immediaetly in all cases.

+

Resolves: #1483

+

ARI: The bridges play and record APIs now handle sample rates > 8K correctly.

+

Author: George Joseph + Date: 2025-09-25

+

The bridge play and record APIs were forcing the Announcer/Recorder channel + to slin8 which meant that if you played or recorded audio with a sample + rate > 8K, it was downsampled to 8K limiting the bandwidth.

+ +

Resolves: #1479

+

DeveloperNote: The ARI /bridges/play and /bridges/record REST APIs have new + parameters that allow the caller to specify the format to be used on the + "Announcer" and "Recorder" channels respecitvely.

+

res_pjsip_geolocation: Add support for Geolocation loc-src parameter

+

Author: Max Grobecker + Date: 2025-09-21

+

This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation. + The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header, + but that option had no effect as it was not implemented by res_pjsip_geolocation.

+

If the location_source configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).

+

This commits adds already documented functionality.

+

sorcery: Move from threadpool to taskpool.

+

Author: Joshua C. Colp + Date: 2025-09-23

+

This change moves observer invocation from the use of + a threadpool to a taskpool. The taskpool options have also + been adjusted to ensure that at least one taskprocessor + remains available at all times.

+

stasis_channels.c: Make protocol_id optional to enable blind transfer via ari

+

Author: Sven Kube + Date: 2025-09-22

+

When handling SIP transfers via ARI, there is no protocol_id in case of + a blind transfer.

+

Resolves: #1467

+

Fix some doxygen, typos and whitespace

+

Author: Bastian Triller + Date: 2025-09-21

+

stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create

+

Author: Sven Kube + Date: 2025-09-18

+

When handling SIP transfers via ARI, the referred_by field in + transfer_ari_state may be null, since SIP REFER requests are not + required to include a Referred-By header. Without this check, a null + value caused the transfer to fail and triggered a NOTIFY with a 500 + Internal Server Error.

+

app_queue: Add NULL pointer checks in app_queue

+

Author: phoneben + Date: 2025-09-11

+

Add NULL check for word_list before calling word_in_list() + Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()

+

Resolves: #1425

+

app_externalivr: Prevent out-of-bounds read during argument processing.

+

Author: Sean Bright + Date: 2025-09-17

+

Resolves: #1422

+

chan_dahdi: Add DAHDI_CHANNEL function.

+

Author: Naveen Albert + Date: 2025-09-11

+

Add a dialplan function that can be used to get/set properties of + DAHDI channels (as opposed to Asterisk channels). This exposes + properties that were not previously available, allowing for certain + operations to now be performed in the dialplan.

+

Resolves: #1455

+

UserNote: The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan.

+

taskpool: Update versions for taskpool stasis options.

+

Author: Joshua C. Colp + Date: 2025-09-16

+

taskpool: Add taskpool API, switch Stasis to using it.

+

Author: Joshua C. Colp + Date: 2025-08-06

+

This change introduces a new API called taskpool. This is a pool + of taskprocessors. It provides the following functionality:

+
    +
  1. Task pushing to a pool of taskprocessors
    2. +
  2. Synchronous tasks
    3. +
  3. Serializers for execution ordering of tasks
    4. +
  4. Growing/shrinking of number of taskprocessors in pool
    5. +
+

This functionality already exists through the combination of + threadpool+taskprocessors but through investigating I determined + that this carries substantial overhead for short to medium duration + tasks. The threadpool uses a single queue of work, and for management + of threads it involves additional tasks.

+

I wrote taskpool to eliminate the extra overhead and management + as much as possible. Instead of a single queue of work each + taskprocessor has its own queue and at push time a selector chooses + the taskprocessor to queue the task to. Each taskprocessor also + has its own thread like normal. This spreads out the tasks immediately + and reduces contention on shared resources.

+

Using the included efficiency tests the number of tasks that can be + executed per second in a taskpool is 6-12 times more than an equivalent + threadpool+taskprocessor setup.

+

Stasis has been moved over to using this new API as it is a heavy consumer + of threadpool+taskprocessors and produces a lot of tasks.

+

UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated + though they continue to be read and used. They have been replaced with taskpool + options that give greater control over the underlying taskpool used for stasis.

+

DeveloperNote: The taskpool API has been added for common usage of a + pool of taskprocessors. It is suggested to use this API instead of the + threadpool+taskprocessor approach.

+

app_adsiprog: Fix possible NULL dereference.

+

Author: Naveen Albert + Date: 2025-09-10

+

get_token can return NULL, but process_token uses this result without + checking for NULL; as elsewhere, check for a NULL result to avoid + possible NULL dereference.

+

Resolves: #1419

+

manager.c: Fix presencestate object leak

+

Author: Nathan Monfils + Date: 2025-09-08

+

ast_presence_state allocates subtype and message. We straightforwardly + need to clean those up.

+

audiohook.c: Ensure correct AO2 reference is dereffed.

+

Author: Sean Bright + Date: 2025-09-10

+

Part of #1440.

+

res_cliexec: Remove unnecessary casts to char*.

+

Author: Naveen Albert + Date: 2025-09-09

+

Resolves: #1436

+

rtp_engine.c: Add exception for comfort noise payload.

+

Author: Ben Ford + Date: 2025-09-09

+

In a previous commit, a change was made to + ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample + rates. This ended up returning an invalid payload int for comfort noise. + A check has been added that returns early if the payload is in fact + supposed to be comfort noise.

+

Fixes: #1340

+

pbx_variables.c: Create real channel for "dialplan eval function".

+

Author: Naveen Albert + Date: 2025-09-09

+

"dialplan eval function" has been using a dummy channel for function + evaluation, much like many of the unit tests. However, sometimes, this + can cause issues for functions that are not expecting dummy channels. + As an example, ast_channel_tech(chan) is NULL on such channels, and + ast_channel_tech(chan)->type consequently results in a NULL dereference. + Normally, functions do not worry about this since channels executing + dialplan aren't dummy channels.

+

While some functions are better about checking for these sorts of edge + cases, use a real channel with a dummy technology to make this CLI + command inherently safe for any dialplan function that could be evaluated + from the CLI.

+

Resolves: #1434

+

res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.

+

Author: George Joseph + Date: 2025-09-23

+

In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets + needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when + AST_RTP_INSTANCE_RTCP_MUX is set.

+

Resolves: #1474

+

chan_websocket: Fix codec validation and add passthrough option.

+

Author: George Joseph + Date: 2025-09-17

+ +

Resolves: #1462

+

res_ari: Ensure outbound websocket config has a websocket_client_id.

+

Author: George Joseph + Date: 2025-09-12

+

Added a check to outbound_websocket_apply() that makes sure an outbound + websocket config object in ari.conf has a websocket_client_id parameter.

+

Resolves: #1457

+

chan_websocket.c: Add DTMF messages

+

Author: Joe Garlick + Date: 2025-09-04

+

Added DTMF messages to the chan_websocket feature.

+

When a user presses DTMF during a call over chan_websocket it will send a message like: + "DTMF_END digit:1"

+

Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70

+

app_queue.c: Add new global 'log_unpause_on_reason_change'

+

Author: Igor Goncharovsky + Date: 2025-09-02

+

In many asterisk-based systems, the pause reason is used to separate + pauses by type,and logically, changing the reason defines two intervals + that should be accounted for separately. The introduction of a new + option allows me to separate the intervals of operator inactivity in + the log by the event of unpausing.

+

UserNote: Add new global option 'log_unpause_on_reason_change' that + is default disabled. When enabled cause addition of UNPAUSE event on + every re-PAUSE with reason changed.

+

app_waitforsilence.c: Use milliseconds to calculate timeout time

+

Author: Igor Goncharovsky + Date: 2025-09-04

+

The functions WaitForNoise() and WaitForSilence() use the time() + functions to calculate elapsed time, which causes the timer to fire on + a whole second boundary, and the actual function execution time to fire + the timer may be 1 second less than expected. This fix replaces time() + with ast_tvnow().

+

Fixes: #1401

+

Fix missing ast_test_flag64 in extconf.c

+

Author: Artem Umerov + Date: 2025-08-29

+

Fix missing ast_test_flag64 after https://github.com/asterisk/asterisk/commit/43bf8a4ded7a65203b766b91eaf8331a600e9d8d

+

pbx_builtins: Allow custom tone for WaitExten.

+

Author: Naveen Albert + Date: 2025-08-25

+

Currently, the 'd' option will play dial tone while waiting + for digits. Allow it to accept an argument for any tone from + indications.conf.

+

Resolves: #1396

+

UserNote: The tone used while waiting for digits in WaitExten + can now be overridden by specifying an argument for the 'd' + option.

+

res_tonedetect: Add option for TONE_DETECT detection to auto stop.

+

Author: Naveen Albert + Date: 2025-08-22

+

One of the problems with TONE_DETECT as it was originally written + is that if a tone is detected multiple times, it can trigger + the redirect logic multiple times as well. For example, if we + do an async goto in the dialplan after detecting a tone, because + the detector is still active until explicitly disabled, if we + detect the tone again, we will branch again and start executing + that dialplan a second time. This is rarely ever desired behavior, + and can happen if the detector is not removed quickly enough.

+

Add a new option, 'e', which automatically disables the detector + once the desired number of matches have been heard. This eliminates + the potential race condition where previously the detector would + need to be disabled immediately, but doing so quickly enough + was not guaranteed. This also allows match criteria to be retained + longer if needed, so the detector does not need to be destroyed + prematurely.

+

Resolves: #1390

+

UserNote: The 'e' option for TONE_DETECT now allows detection to + be disabled automatically once the desired number of matches have + been fulfilled, which can help prevent race conditions in the + dialplan, since TONE_DETECT does not need to be disabled after + a hit.

+

app_queue: fix comparison for announce-position-only-up

+

Author: Stuart Henderson + Date: 2025-08-21

+

Numerically comparing that the current queue position is less than + last_pos_said can only be done after at least one announcement has been + made, otherwise last_pos_said is at the default (0).

+

Fixes: #1386

+

sorcery: Prevent duplicate objects and ensure missing objects are created on update

+

Author: Alexei Gradinari + Date: 2025-07-07

+

This patch resolves two issues in Sorcery objectset handling with multiple + backends:

+
    +
  1. +

    Prevent duplicate objects: + When an object exists in more than one backend (e.g., a contact in both + 'astdb' and 'realtime'), the objectset previously returned multiple instances + of the same logical object. This caused logic failures in components like the + PJSIP registrar, where duplicate contact entries led to overcounting and + incorrect deletions, when max_contacts=1 and remove_existing=yes.

    +

    This patch ensures only one instance of an object with a given key is added + to the objectset, avoiding these duplicate-related side effects.

    +
    2. +
  2. +

    Ensure missing objects are created: + When using multiple writable backends, a temporary backend failure can lead + to objects missing permanently from that backend. + Currently, .update() silently fails if the object is not present, + and no .create() is attempted. + This results in inconsistent state across backends (e.g. astdb vs. realtime).

    +

    This patch introduces a new global option in sorcery.conf: + [general] + update_or_create_on_update_miss = yes|no

    +

    Default: no (preserves existing behavior).

    +

    When enabled: if .update() fails with no data found, .create() is attempted + in that backend. This ensures that objects missing due to temporary backend + outages are re-synchronized once the backend is available again.

    +

    Added a new CLI command: + sorcery show settings + Displays global Sorcery settings, including the current value of + update_or_create_on_update_miss.

    +

    Updated tests to validate both flag enabled/disabled behavior.

    +
    3. +
+

Fixes: #1289

+

UserNote: Users relying on Sorcery multiple writable backends configurations + (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes + in sorcery.conf to ensure missing objects are recreated after temporary backend + failures. Default behavior remains unchanged unless explicitly enabled.

+

sig_analog: Skip Caller ID spill if usecallerid=no.

+

Author: Naveen Albert + Date: 2025-08-25

+

If Caller ID is disabled for an FXS port, then we should not send any + Caller ID spill on the line, as we have no Caller ID information that + we can/should be sending.

+

Resolves: #1394

+

chan_dahdi: Fix erroneously persistent dialmode.

+

Author: Naveen Albert + Date: 2025-08-18

+

It is possible to modify the dialmode setting in the chan_dahdi/sig_analog + private using the CHANNEL function, to modify it during calls. However, + it was not being reset between calls, meaning that if, for example, tone + dialing was disabled, it would never work again unless explicitly enabled.

+

This fixes the setting by pairing it with a "perm" version of the setting, + as a few other features have, so that it can be reset to the permanent + setting between calls. The documentation is also clarified to explain + the interaction of this setting and the digitdetect setting more clearly.

+

Resolves: #1378

+

chan_websocket: Allow additional URI parameters to be added to the outgoing URI.

+

Author: George Joseph + Date: 2025-08-13

+ +

Resolves: #1352

+

UserNote: A new WebSocket channel driver option v has been added to the + Dial application that allows you to specify additional URI parameters on + outgoing connections. Run core show application Dial from the Asterisk CLI + to see how to use it.

+

chan_websocket: Fix buffer overrun when processing TEXT websocket frames.

+

Author: George Joseph + Date: 2025-08-19

+

ast_websocket_read() receives data into a fixed 64K buffer then continually + reallocates a final buffer that, after all continuation frames have been + received, is the exact length of the data received and returns that to the + caller. process_text_message() in chan_websocket was attempting to set a + NULL terminator on the received payload assuming the payload buffer it + received was the large 64K buffer. The assumption was incorrect so when it + tried to set a NULL terminator on the payload, it could, depending on the + state of the heap at the time, cause heap corruption.

+

process_text_message() now allocates its own payload_len + 1 sized buffer, + copies the payload received from ast_websocket_read() into it then NULL + terminates it prevent the possibility of the overrun and corruption.

+

Resolves: #1384

+

sig_analog: Fix SEGV due to calling strcmp on NULL.

+

Author: Naveen Albert + Date: 2025-08-18

+

Add an additional check to guard against the channel application being + NULL.

+

Resolves: #1380

+

ARI: Add command to indicate progress to a channel

+

Author: Sven Kube + Date: 2025-07-30

+

Adds an ARI command to send a progress indication to a channel.

+

DeveloperNote: A new ARI endpoint is available at /channels/{channelId}/progress to indicate progress to a channel.

+

dsp.c: Improve debug logging in tone_detect().

+

Author: Naveen Albert + Date: 2025-08-15

+

The debug logging during DSP processing has always been kind + of overwhelming and annoying to troubleshoot. Simplify and + improve the logging in a few ways to aid DSP debugging:

+ +

Resolves: #1375

+

res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.

+

Author: Jose Lopes + Date: 2025-07-30

+

After an asterisk restart, the deletion of ARI Devicestates didn't + return error, but the devicestate was not deleted. + Found a typo on populate_cache function that created wrong cache for + device states. + This bug caused wrong assumption that devicestate didn't exist, + since it was not in cache, so deletion didn't returned error.

+

Fixes: #1327

+

app_chanspy: Add option to not automatically answer channel.

+

Author: Naveen Albert + Date: 2025-08-13

+

Add an option for ChanSpy and ExtenSpy to not answer the channel + automatically. Most applications that auto-answer by default + already have an option to disable this behavior if unwanted.

+

Resolves: #1358

+

UserNote: ChanSpy and ExtenSpy can now be configured to not + automatically answer the channel by using the 'N' option.

+

xmldoc.c: Fix rendering of CLI output.

+

Author: George Joseph + Date: 2025-08-14

+

If you do a core show application Dial, you'll see it's kind of a mess. + Indents are wrong is some places, examples are printed in black which makes + them invisible on most terminals, and the lack of line breaks in some cases + makes it hard to follow.

+ +

Example from Dial before fixes: + ``` + Example: Dial 555-1212 on first available channel in group 1, searching + from highest to lowest

+
  Example: Ringing FXS channel 4 with ring cadence 2
+
+  Example: Dial 555-1212 on channel 3 and require answer confirmation
+
+

...

+
  O([mode]):
+      mode - With <mode> either not specified or set to '1', the originator
+      hanging up will cause the phone to ring back immediately.
+
+ +

After: + ``` + Example: Dial 555-1212 on first available channel in group 1, searching + from highest to lowest

+
   same => n,Dial(DAHDI/g1/5551212)
+
+  Example: Ringing FXS channel 4 with ring cadence 2
+
+   same => n,Dial(DAHDI/4r2)
+
+  Example: Dial 555-1212 on channel 3 and require answer confirmation
+
+   same => n,Dial(DAHDI/3c/5551212)
+
+

...

+
  O([mode]):
+      mode - With <mode> either not specified or set to '1', the originator
+      hanging up will cause the phone to ring back immediately.
+      With <mode> set to '2', when the operator flashes the trunk, it will
+      ring their phone back.
+      Enables *operator services* mode.  This option only works when bridging
+      a DAHDI channel to another DAHDI channel only. If specified on
+      non-DAHDI interfaces, it will be ignored. When the destination answers
+      (presumably an operator services station), the originator no longer has
+      control of their line. They may hang up, but the switch will not
+      release their line until the destination party (the operator) hangs up.
+
+  p:
+      This option enables screening mode. This is basically Privacy mode
+      without memory.
+
+

```

+

There are still things we can do to make this more readable but this is a + start.

+

func_frame_drop: Add debug messages for dropped frames.

+

Author: Naveen Albert + Date: 2025-08-14

+

Add debug messages in scenarios where frames that are usually processed + are dropped or skipped.

+

Resolves: #1371

+

test_res_prometheus: Fix compilation failure on Debian 13.

+

Author: Naveen Albert + Date: 2025-08-14

+

curl_easy_setopt expects long types, so be explicit.

+

Resolves: #1369

+

func_frame_drop: Handle allocation failure properly.

+

Author: Naveen Albert + Date: 2025-08-14

+

Handle allocation failure and simplify the allocation using asprintf.

+

Resolves: #1366

+

pbx_lua.c: segfault when pass null data to term_color function

+

Author: Alexey Khabulyak + Date: 2025-08-14

+

This can be reproduced under certain curcomstences. + For example: call app.playback from lua with invalid data: app.playback({}). + pbx_lua.c will try to get data for this playback using lua_tostring function. + This function returs NULL for everything but strings and numbers. + Then, it calls term_color with NULL data. + term_color function can call(if we don't use vt100 compat term) + ast_copy_string with NULL inbuf which cause segfault. bt example: + ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0) + at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412

+

Resolves: https://github.com/asterisk/asterisk/issues/1363

+

bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.

+

Author: Naveen Albert + Date: 2025-08-14

+

If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed + to answer when another channel bridges to it using Bridge(), and this is + checked when ast_bridge_call* is called. However, another path exists + (bridge_exec -> ast_bridge_add_channel) where this variable was not + checked and channels would be answered. We now check the variable there.

+

Resolves: #401 + Resolves: #1364

+

res_rtp_asterisk: Don't send RTP before DTLS has negotiated.

+

Author: Ben Ford + Date: 2025-08-04

+

There was no check in __rtp_sendto that prevented Asterisk from sending + RTP before DTLS had finished negotiating. This patch adds logic to do + so.

+

Fixes: #1260

+

app_dial.c: Moved channel lock to prevent deadlock

+

Author: Alexey Khabulyak + Date: 2025-08-04

+

It's reproducible with pbx_lua, not regular dialplan.

+

deadlock description: + 1. asterisk locks a channel + 2. calls function onedigit_goto + 3. calls ast_goto_if_exists funciton + 4. checks ast_exists_extension -> pbx_extension_helper + 5. pbx_extension_helper calls pbx_find_extension + 6. Then asterisk starts autoservice in a new thread + 7. autoservice run tries to lock the channel again

+

Because our channel is locked already, autoservice can't lock. + Autoservice can't lock -> autoservice stop is waiting forever. + onedigit_goto waits for autoservice stop.

+

Resolves: https://github.com/asterisk/asterisk/issues/1335

+

res_pjsip_diversion: resolve race condition between Diversion header processing and redirect

+

Author: Mike Bradeen + Date: 2025-08-07

+

Based on the firing order of the PJSIP call-backs on a redirect, it was possible for + the Diversion header to not be included in the outgoing 181 response to the UAC and + the INVITE to the UAS.

+

This change moves the Diversion header processing to an earlier PJSIP callback while also + preventing the corresponding update that can cause a duplicate 181 response when processing + the header at that time.

+

Resolves: #1349

+

file.c: with "sounds_search_custom_dir = yes", search "custom" directory

+

Author: Allan Nathanson + Date: 2025-08-10

+

With sounds_search_custom_dir = yes, we are supposed to search for sounds + in the AST_DATA_DIR/sounds/custom directory before searching the normal + directories. Unfortunately, a recent change + (https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in + the "custom" directory not being searched. This change restores this + expected behavior.

+

Resolves: #1353

+

cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.

+

Author: Sperl Viktor + Date: 2025-06-30

+

Fixes: #1280

+

UserNote: Enabling the tracking of the + STREAM_BEGIN and the STREAM_END event + types in cel.conf will log media files and + music on hold played to each channel. + The STREAM_BEGIN event's extra field will + contain a JSON with the file details (path, + format and language), or the class name, in + case of music on hold is played. The DTMF + event's extra field will contain a JSON with + the digit and the duration in milliseconds.

+

channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.

+

Author: George Joseph + Date: 2025-07-30

+

The fact that deleting an object from a map invalidates any iterator + that happens to currently point to that object was overlooked in the initial + implementation. Unfortunately, there's no way to detect that an iterator + has been invalidated so the result was an occasional SEGV triggered by modules + like app_chanspy that opens an iterator and can keep it open for a long period + of time. The new implementation doesn't keep the underlying C++ iterator + open across calls to ast_channel_iterator_next() and uses a read lock + on the map to ensure that, even for the few microseconds we use the + iterator, another thread can't delete a channel from under it. Even with + this change, the iterators are still WAY faster than the ao2_legacy + storage driver.

+

Full details about the new implementation are located in the comments for + iterator_next() in channelstorage_cpp_map_name_id.cc.

+

Resolves: #1309

+

res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM

+

Author: George Joseph + Date: 2025-08-05

+

UserNote: Options are now available in the menuselect "Resource Modules" + category that allow you to enable the AES_192, AES_256 and AES_GCM + cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support + them but modern versions do. Previously, the only way to enable them was + to set the CFLAGS environment variable when running ./configure. + The default setting is to disable them preserving existing behavior.

+

cdr: add CANCEL dispostion in CDR

+

Author: zhou_jiajian + Date: 2025-07-24

+

In the original implementation, both CANCEL and NO ANSWER states were + consolidated under the NO ANSWER disposition. This patch introduces a + separate CANCEL disposition, with an optional configuration switch to + enable this new disposition.

+

Resolves: #1323

+

UserNote: A new CDR option "canceldispositionenabled" has been added + that when set to true, the NO ANSWER disposition will be split into + two dispositions: CANCEL and NO ANSWER. The default value is 'no'

+

func_curl: Allow auth methods to be set.

+

Author: Naveen Albert + Date: 2025-08-01

+

Currently the CURL function only supports Basic Authentication, + the default auth method in libcurl. Add an option that also + allows enabling digest authentication.

+

Resolves: #1332

+

UserNote: The httpauth field in CURLOPT now allows the authentication + methods to be set.

+

options: Change ast_options from ast_flags to ast_flags64.

+

Author: George Joseph + Date: 2025-07-21

+

DeveloperNote: The 32-bit ast_options has no room left to accomodate new + options and so has been converted to an ast_flags64 structure. All internal + references to ast_options have been updated to use the 64-bit flag + manipulation macros. External module references to the 32-bit ast_options + should continue to work on little-endian systems because the + least-significant bytes of a 64 bit integer will be in the same location as a + 32-bit integer. Because that's not the case on big-endian systems, we've + swapped the bytes in the flags manupulation macros on big-endian systems + so external modules should still work however you are encouraged to test.

+

res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)

+

Author: Alexei Gradinari + Date: 2025-07-15

+

This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly + falls back to the next configured backend when the current one returns + SQL_NO_DATA (i.e., no record found). + This is a logical error and performance risk in multi-backend configurations.

+

Solution: + Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker. + ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found. + Core engine stops iterating on this marker, avoiding unnecessary fallback.

+

Notes: + Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly. + This patch only covers ODBC.

+

Fixes: #1305

+

resource_channels.c: Don't call ast_channel_get_by_name on empty optional arguments

+

Author: Sven Kube + Date: 2025-07-30

+

ast_ari_channels_create and ast_ari_channels_dial called the + ast_channel_get_by_name function with optional arguments. Since + 8f1982c4d6, this function logs an error for empty channel names. + This commit adds checks for empty optional arguments that are used + to call ast_channel_get_by_name to prevent these error logs.

+

app_agent_pool: Remove documentation for removed option.

+

Author: Naveen Albert + Date: 2025-07-28

+

The already-deprecated "password" option for the AGENT function was + removed in commit d43b17a872e8227aa8a9905a21f90bd48f9d5348 for + Asterisk 12, but the documentation for it wasn't removed then.

+

Resolves: #1321

+

pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper should return directly.

+

Author: Tinet-mucw + Date: 2025-07-22

+

Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. + In the situation that action Redirect is broken by pbx_extension_helper this info is changed. + This will cause the current dialplan location to be executed twice. + In other words, the Redirect action does not take effect.

+

Resolves: #1315

+

res_agi: Increase AGI command buffer size from 2K to 8K

+

Author: Sperl Viktor + Date: 2025-07-22

+

Fixes: #1317

+

ast_tls_cert: Make certificate validity configurable.

+

Author: Naveen Albert + Date: 2025-07-16

+

Currently, the ast_tls_cert script is hardcoded to produce certificates + with a validity of 365 days, which is not generally desirable for self- + signed certificates. Make this parameter configurable.

+

Resolves: #1307

+

cdr.c: Set tenantid from party_a->base instead of chan->base.

+

Author: George Joseph + Date: 2025-07-17

+

The CDR tenantid was being set in cdr_object_alloc from the channel->base + snapshot. Since this happens at channel creation before the dialplan is even + reached, calls to CHANNEL(tenantid)=<something> in the dialplan were being + ignored. Instead we now take tenantid from party_a when + cdr_object_create_public_records() is called which is after the call has + ended and all channel snapshots rebuilt. This is exactly how accountcode + and amaflags, which can also be set in tha dialplpan, are handled.

+

Resolves: #1259

+

app_mixmonitor: Update the documentation concerning the "D" option.

+

Author: George Joseph + Date: 2025-07-16

+

When using the "D" option to output interleaved audio, the file extension + must be ".raw". That info wasn't being properly rendered in the markdown + and HTML on the documentation site. The XML was updated to move the + note in the option section to a warning in the description.

+

Resolves: #1269

+

sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.

+

Author: Naveen Albert + Date: 2025-07-14

+

Previously, we were only using # (ST) as a terminator, and not handling + A (STP), B (ST2P), or C (ST3P), which erroneously led to it being + treated as part of the dialed number. Parse any of these as the start + digit.

+

Resolves: #1301

+

chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA

+

Author: kodokaii + Date: 2025-07-03

+

In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from + the queue but does not reset the frame_queue_length counter.

+

As a result, the driver incorrectly thinks the queue is full after flushing, + which prevents new multimedia frames from being sent, especially after multiple + flush commands.

+

This fix sets frame_queue_length to 0 after flushing, ensuring the queue state + is consistent with its actual content.

+

Fixes: #1304

+

chan_pjsip.c: Change SSRC after media source change

+

Author: Martin Tomec + Date: 2025-06-25

+

When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly. + To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream. + This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.

+

Resolves: #927

+

Media over Websocket Channel Driver

+

Author: George Joseph + Date: 2025-04-28

+ +

UserNote: A new channel driver "chan_websocket" is now available. It can + exchange media over both inbound and outbound websockets and will both frame + and re-time the media it receives. + See http://s.asterisk.net/mow for more information.

+

UserNote: The ARI channels/externalMedia API now includes support for the + WebSocket transport provided by chan_websocket.

+

bundled_pjproject: Avoid deadlock between transport and transaction

+

Author: Stanislav Abramenkov + Date: 2025-07-01

+

Backport patch from upstream + * Avoid deadlock between transport and transaction + https://github.com/pjsip/pjproject/commit/edde06f261ac

+

Issue described in + https://github.com/pjsip/pjproject/issues/4442

+

utils.h: Add rounding to float conversion to int.

+

Author: mkmer + Date: 2025-03-23

+

Quote from an audio engineer NR9V: + There is a minor issue of a small amount of crossover distortion though as a result of ast_slinear_saturated_multiply_float() not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion.

+

Resolves: #1176

+

pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return -1.

+

Author: Tinet-mucw + Date: 2025-06-18

+

Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. + In the situation that action Redirect is broken by GotoIf this info is changed. + that will causes confusion in dialplan execution.

+

Resolves: #1273

+

res_musiconhold.c: Ensure we're always locked around music state access.

+

Author: Sean Bright + Date: 2025-04-08

+

res_musiconhold.c: Annotate when the channel is locked.

+

Author: Sean Bright + Date: 2025-04-08

+

res_musiconhold: Appropriately lock channel during start.

+

Author: Jaco Kroon + Date: 2024-12-19

+

This relates to #829

+

This doesn't sully solve the Ops issue, but it solves the specific crash + there. Further PRs to follow.

+

In the specific crash the generator was still under construction when + moh was being stopped, which then proceeded to close the stream whilst + it was still in use.

+

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

+

res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.

+

Author: George Joseph + Date: 2025-08-28

+

In the highly-unlikely event that get_authorization_hdr() couldn't find an + Authorization header in a request, trying to get the digest algorithm + would cauase a SEGV. We now check that we have an auth header that matches + the realm before trying to get the algorithm from it.

+

Resolves: #GHSA-64qc-9x89-rx5j

+

safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.

+

Author: ThatTotallyRealMyth + Date: 2025-06-10

+

UpgradeNote: The safe_asterisk script now checks that, if it was run by the + root user, the /etc/asterisk/startup.d directory and all the files it contains + are owned by root. If the checks fail, safe_asterisk will exit with an error + and Asterisk will not be started. Additionally, the default logging + destination is now stderr instead of tty "9" which probably won't exist + in modern systems.

+

Resolves: #GHSA-v9q8-9j8m-5xwp

+

res_stir_shaken: Test for missing semicolon in Identity header.

+

Author: George Joseph + Date: 2025-07-31

+

ast_stir_shaken_vs_verify() now makes sure there's a semicolon in + the Identity header to prevent a possible segfault.

+

Resolves: #GHSA-mrq5-74j5-f5cr

+

channelstorage: Rename callbacks that conflict with DEBUG_FD_LEAKS.

+

Author: George Joseph + Date: 2025-07-08

+

DEBUG_FD_LEAKS replaces calls to "open" and "close" with functions that keep + track of file descriptors, even when those calls are actually callbacks + defined in structures like ast_channelstorage_instance->open and don't touch + file descriptors. This causes compilation failures. Those callbacks + have been renamed to "open_instance" and "close_instance" respectively.

+

Resolves: #1287

+

channelstorage_cpp_map_name_id: Fix callback returning non-matching channels.

+

Author: George Joseph + Date: 2025-07-09

+

When the callback() API was invoked but no channel passed the test, callback + would return the last channel tested instead of NULL. It now correctly + returns NULL when no channel matches.

+

Resolves: #1288

+

audiohook.c: Improve frame pairing logic to avoid MixMonitor breakage with mixed codecs

+

Author: Michal Hajek + Date: 2025-05-21

+

This patch adjusts the read/write synchronization logic in audiohook_read_frame_both() + to better handle calls where participants use different codecs or sample sizes + (e.g., alaw vs G.722). The previous hard threshold of 2 * samples caused MixMonitor + recordings to break or stutter when frames were not aligned between both directions.

+

The new logic uses a more tolerant limit (1.5 * samples), which prevents audio tearing + without causing excessive buffer overruns. This fix specifically addresses issues + with MixMonitor when recording directly on a channel in a bridge using mixed codecs.

+

Reported-by: Michal Hajek michal.hajek@daktela.com

+

Resolves: #1276 + Resolves: #1279

+

channelstorage_makeopts.xml: Remove errant XML character.

+

Author: Sean Bright + Date: 2025-06-30

+

Resolves: #1282

+

res_stir_shaken.so: Handle X5U certificate chains.

+

Author: George Joseph + Date: 2025-06-18

+

The verification process will now load a full certificate chain retrieved + via the X5U URL instead of loading only the end user cert.

+ +

Resolves: #1272

+

UserNote: The STIR/SHAKEN verification process will now load a full + certificate chain retrieved via the X5U URL instead of loading only + the end user cert.

+

res_stir_shaken: Add "ignore_sip_date_header" config option.

+

Author: George Joseph + Date: 2025-06-15

+

UserNote: A new STIR/SHAKEN verification option "ignore_sip_date_header" has + been added that when set to true, will cause the verification process to + not consider a missing or invalid SIP "Date" header to be a failure. This + will make the IAT the sole "truth" for Date in the verification process. + The option can be set in the "verification" and "profile" sections of + stir_shaken.conf.

+

Also fixed a bug in the port match logic.

+

Resolves: #1251 + Resolves: #1271

+

app_record: Add RECORDING_INFO function.

+

Author: Naveen Albert + Date: 2024-01-22

+

Add a function that can be used to retrieve info + about a previous recording, such as its duration.

+

This is being added as a function to avoid possibly + trampling on dialplan variables, and could be extended + to provide other information in the future.

+

Resolves: #548

+

UserNote: The RECORDING_INFO function can now be used + to retrieve the duration of a recording.

+

app_sms.c: Fix sending and receiving SMS messages in protocol 2

+

Author: Itzanh + Date: 2025-04-06

+

This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2.

+ +

One of the bugs caused messages to have random characters and junk appended at the end up to the character limit. Another bug prevented Asterisk from sending messages from Asterisk to the phone at all. A final bug caused the transmission from Asterisk to the phone to take a long time because app_sms.c did not hang up after correctly sending the message, causing the phone to have to time out and hang up in order to complete the message transmission.

+

This was tested with a Linksys PAP2T and with a GrandStream HT814, sending and receiving messages with Telefónica DOMO Mensajes phones from Telefónica Spain. I had to play with both the network jitter buffer and the dB gain to get it to work. One of my phones required the gain to be set to +3dB for it to work, while another required it to be set to +6dB.

+

Only MORX and MTTX were tested, I did not test sending and receiving messages to a TelCo SMSC.

+

app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range

+

Author: phoneben + Date: 2025-05-26

+

This update adds support for a new QUEUE_RAISE_PENALTY format: rN

+

When QUEUE_RAISE_PENALTY is set to rN (e.g., r4), only members whose current penalty + is greater than or equal to the defined min_penalty and less than or equal to max_penalty + will have their penalty raised to N.

+

Members with penalties outside the min/max range remain unchanged.

+

Example behaviors:

+

QUEUE_RAISE_PENALTY=4 → Raise all members with penalty < 4 (existing behavior) + QUEUE_RAISE_PENALTY=r4 → Raise only members with penalty in [min_penalty, max_penalty] to 4

+

Implementation details:

+

Adds parsing logic to detect the r prefix and sets the raise_respect_min flag

+

Modifies the raise logic to skip members outside the defined penalty range when the flag is active

+

UserNote: This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises + only for members whose current penalty is within the [min_penalty, max_penalty] range. + Members with lower or higher penalties are unaffected. + This behavior is backward-compatible with existing queue rule configurations.

+

res_websocket_client: Add more info to the XML documentation.

+

Author: George Joseph + Date: 2025-06-05

+

Added "see-also" links to chan_websocket and ARI Outbound WebSocket and + added an example configuration for each.

+

res_odbc: cache_size option to limit the cached connections.

+

Author: Jaco Kroon + Date: 2024-12-13

+

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

+

UserNote: New cache_size option for res_odbc to on a per class basis limit the + number of cached connections. Please reference the sample configuration + for details.

+

res_odbc: cache_type option for res_odbc.

+

Author: Jaco Kroon + Date: 2024-12-10

+

This enables setting cache_type classes to a round-robin queueing system + rather than the historic stack mechanism.

+

This should result in lower risk of connection drops due to shorter idle + times (the first connection to go onto the stack could in theory never + be used again, ever, but sit there consuming resources, there could be + multiple of these).

+

And with a queue rather than a stack, dead connections are guaranteed to + be detected and purged eventually.

+

This should end up better balancing connection_cnt with actual load + over time, assuming the database doesn't keep connections open + excessively long from it's side.

+

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

+

UserNote: When using res_odbc it should be noted that back-end + connections to the underlying database can now be configured to re-use + the cached connections in a round-robin manner rather than repeatedly + re-using the same connection. This helps to keep connections alive, and + to purge dead connections from the system, thus more dynamically + adjusting to actual load. The downside is that one could keep too many + connections active for a longer time resulting in resource also begin + consumed on the database side.

+

res_pjsip: Fix empty ActiveChannels property in AMI responses.

+

Author: Sean Bright + Date: 2025-05-27

+

The logic appears to have been reversed since it was introduced in + 05cbf8df.

+

Resolves: #1254

+

ARI Outbound Websockets

+

Author: George Joseph + Date: 2025-03-28

+

Asterisk can now establish websocket sessions to your ARI applications + as well as accepting websocket sessions from them. + Full details: http://s.asterisk.net/ari-outbound-ws

+

Code change summary: + * Added an ast_vector_string_join() function, + * Added ApplicationRegistered and ApplicationUnregistered ARI events. + * Converted res/ari/config.c to use sorcery to process ari.conf. + * Added the "outbound-websocket" ARI config object. + * Refactored res/ari/ari_websockets.c to handle outbound websockets. + * Refactored res/ari/cli.c for the sorcery changeover. + * Updated res/res_stasis.c for the sorcery changeover. + * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. + * Added CLI commands to manage ARI websockets. + * Added the new "outbound-websocket" object to ari.conf.sample. + * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml

+

UserNote: Asterisk can now establish websocket sessions to your ARI applications + as well as accepting websocket sessions from them. + Full details: http://s.asterisk.net/ari-outbound-ws

+

res_websocket_client: Create common utilities for websocket clients.

+

Author: George Joseph + Date: 2025-05-02

+

Since multiple Asterisk capabilities now need to create websocket clients + it makes sense to create a common set of utilities rather than making + each of those capabilities implement their own.

+ +

Also as part of thie commit, several sorcery convenience macros were created + to make registering common object fields easier.

+

UserNote: A new module "res_websocket_client" and config file + "websocket_client.conf" have been added to support several upcoming new + capabilities that need common websocket client configuration.

+

asterisk.c: Add option to restrict shell access from remote consoles.

+

Author: George Joseph + Date: 2025-05-19

+

UserNote: A new asterisk.conf option 'disable_remote_console_shell' has + been added that, when set, will prevent remote consoles from executing + shell commands using the '!' prefix.

+

Resolves: #GHSA-c7p6-7mvq-8jq2

+

res_pjsip_messaging.c: Mask control characters in received From display name

+

Author: George Joseph + Date: 2025-03-24

+

Incoming SIP MESSAGEs will now have their From header's display name + sanitized by replacing any characters < 32 (space) with a space.

+

Resolves: #GHSA-2grh-7mhv-fcfw

+

frame.c: validate frame data length is less than samples when adjusting volume

+

Author: mkmer + Date: 2025-05-12

+

Resolves: #1230

+

res_audiosocket.c: Add retry mechanism for reading data from AudioSocket

+

Author: Sven Kube + Date: 2025-05-13

+

The added retry mechanism addresses an issue that arises when fragmented TCP + packets are received, each containing only a portion of an AudioSocket packet. + This situation can occur if the external service sending the AudioSocket data + has Nagle's algorithm enabled.

+

res_audiosocket.c: Set the TCP_NODELAY socket option

+

Author: Sven Kube + Date: 2025-05-13

+

Disable Nagle's algorithm by setting the TCP_NODELAY socket option. + This reduces latency by preventing delays caused by packet buffering.

+

menuselect: Fix GTK menu callbacks for Fedora 42 compatibility

+

Author: Thomas B. Clark + Date: 2025-05-12

+

This patch resolves a build failure in menuselect_gtk.c when running + make menuconfig on Fedora 42. The new version of GTK introduced stricter + type checking for callback signatures.

+

Changes include: + - Add wrapper functions to match the expected void (*)(void) signature. + - Update menu_items array to use these wrappers.

+

Fixes: #1243

+

jansson: Upgrade version to jansson 2.14.1

+

Author: Stanislav Abramenkov + Date: 2025-03-24

+

UpgradeNote: jansson has been upgraded to 2.14.1. For more + information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.14.1

+

Resolves: #1178

+

pjproject: Increase maximum SDP formats and attribute limits

+

Author: Joe Searle + Date: 2025-05-15

+

Since Chrome 136, using Windows, when initiating a video call the INVITE SDP exceeds the maximum number of allowed attributes, resulting in the INVITE being rejected. This increases the attribute limit and the number of formats allowed when using bundled pjproject.

+

Fixes: #1240

+

manager.c: Invalid ref-counting when purging events

+

Author: Nathan Monfils + Date: 2025-05-05

+

We have a use-case where we generate a lot of events on the AMI, and + then when doing manager show eventq we would see some events which + would linger for hours or days in there. Obviously something was leaking. + Testing allowed us to track down this logic bug in the ref-counting on + the event purge.

+

Reproducing the bug was not super trivial, we managed to do it in a + production-like load testing environment with multiple AMI consumers.

+

The race condition itself:

+
    +
  1. something allocates and links session
    2. +
  2. purge_sessions iterates over that session (takes ref)
    3. +
  3. purge_session correctly de-referencess that session
    4. +
  4. purge_session re-evaluates the while() loop, taking a reference
    5. +
  5. purge_session exits (n_max > 0 is false)
    6. +
  6. whatever allocated the session deallocates it, but a reference is + now lost since we exited the while loop before de-referencing.
    7. +
  7. since the destructor is never called, the session->last_ev->usecount + is never decremented, leading to events lingering in the queue
    8. +
+

The impact of this bug does not seem major. The events are small and do + not seem, from our testing, to be causing meaningful additional CPU + usage. Mainly we wanted to fix this issue because we are internally + adding prometheus metrics to the eventq and those leaked events were + causing the metrics to show garbage data.

+

res_pjsip_nat.c: Do not overwrite transfer host

+

Author: Mike Bradeen + Date: 2025-05-08

+

When a call is transfered via dialplan behind a NAT, the + host portion of the Contact header in the 302 will no longer + be over-written with the external NAT IP and will retain the + hostname.

+

Fixes: #1141

+

chan_pjsip: Serialize INVITE creation on DTMF attended transfer

+

Author: Mike Bradeen + Date: 2025-05-05

+

When a call is transfered via DTMF feature code, the Transfer Target and + Transferer are bridged immediately. This opens the possibilty of a race + condition between the creation of an INVITE and the bridge induced colp + update that can result in the set caller ID being over-written with the + transferer's default info.

+

Fixes: #1234

+

Alternate Channel Storage Backends

+

Author: George Joseph + Date: 2024-12-31

+

Full details: http://s.asterisk.net/dc679ec3

+

The previous proof-of-concept showed that the cpp_map_name_id alternate + storage backed performed better than all the others so this final PR + adds only that option. You still need to enable it in menuselect under + the "Alternate Channel Storage Backends" category.

+

To select which one is used at runtime, set the "channel_storage_backend" + option in asterisk.conf to one of the values described in + asterisk.conf.sample. The default remains "ao2_legacy".

+

UpgradeNote: With this release, you can now select an alternate channel + storage backend based on C++ Maps. Using the new backend may increase + performance and reduce the chances of deadlocks on heavily loaded systems. + For more information, see http://s.asterisk.net/dc679ec3

+

sig_analog: Add Call Waiting Deluxe support.

+

Author: Naveen Albert + Date: 2023-08-24

+

Adds support for Call Waiting Deluxe options to enhance + the current call waiting feature.

+

As part of this change, a mechanism is also added that + allows a channel driver to queue an audio file for Dial() + to play, which is necessary for the announcement function.

+

ASTERISK-30373 #close

+

Resolves: #271

+

UserNote: Call Waiting Deluxe can now be enabled for FXS channels + by enabling its corresponding option.

+

app_sms: Ignore false positive vectorization warning.

+

Author: Naveen Albert + Date: 2025-01-24

+

Ignore gcc warning about writing 32 bytes into a region of size 6, + since we check that we don't go out of bounds for each byte. + This is due to a vectorization bug in gcc 15, stemming from + gcc commit 68326d5d1a593dc0bf098c03aac25916168bc5a9.

+

Resolves: #1088

+

lock.h: Add include for string.h when DEBUG_THREADS is defined.

+

Author: George Joseph + Date: 2025-05-02

+

When DEBUG_THREADS is defined, lock.h uses strerror(), which is defined + in the libc string.h file, to print warning messages. If the including + source file doesn't include string.h then strerror() won't be found and + and compile errors will be thrown. Since lock.h depends on this, string.h + is now included from there if DEBUG_THREADS is defined. This way, including + source files don't have to worry about it.

+

res_pjsip_caller_id: Also parse URI parameters for ANI2.

+

Author: Naveen Albert + Date: 2025-04-26

+

If the isup-oli was sent as a URI parameter, rather than a header + parameter, it was not being parsed. Make sure we parse both if + needed so the ANI2 is set regardless of which type of parameter + the isup-oli is sent as.

+

Resolves: #1220

+

app_meetme: Remove inaccurate removal version from xmldocs.

+

Author: Naveen Albert + Date: 2025-04-26

+

app_meetme is deprecated but wasn't removed as planned in 21, + so remove the inaccurate removal version.

+

Resolves: #1224

+

docs: Fix typos in apps/

+

Author: Luz Paz + Date: 2025-04-09

+

Found via codespell

+

stasis/control.c: Set Hangup Cause to No Answer on Dial timeout

+

Author: Mike Bradeen + Date: 2025-04-17

+

Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, + but the Dial command via ARI did not set an explicit reason. This resulted in a + CANCEL with Normal Call Clearing and corresponding ChannelDestroyed.

+

This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the + other operations.

+

Fixes: #963

+

UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.

+

chan_iax2: Minor improvements to documentation and warning messages.

+

Author: Naveen Albert + Date: 2025-04-18

+ +

Resolves: #1206

+

pbx_ael: unregister AELSub application and CLI commands on module load failure

+

Author: Andreas Wehrmann + Date: 2025-04-18

+

This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out, + that the AEL module doesn't do proper cleanup when it fails to load. + This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module() + returns an error but load_module() doesn't then unregister CLI cmds and the application.

+

res_pjproject: Fix DTLS client check failing on some platforms

+

Author: Albrecht Oster + Date: 2025-04-10

+

Certain platforms (mainly BSD derivatives) have an additional length + field in sockaddr_in6 and sockaddr_in. + ast_sockaddr_from_pj_sockaddr() does not take this field into account + when copying over values from the pj_sockaddr into the ast_sockaddr. + The resulting ast_sockaddr will have an uninitialized value for + sin6_len/sin_len while the other ast_sockaddr (not converted from + a pj_sockaddr) to check against in ast_sockaddr_pj_sockaddr_cmp() + has the correct length value set.

+

This has the effect that ast_sockaddr_cmp() will always indicate + an address mismatch, because it does a bitwise comparison, and all DTLS + packets are dropped even if addresses and ports match.

+

ast_sockaddr_from_pj_sockaddr() now checks whether the length fields + are available on the current platform and sets the values accordingly.

+

Resolves: #505

+

Prequisites for ARI Outbound Websockets

+

Author: George Joseph + Date: 2025-04-16

+

stasis: + * Added stasis_app_is_registered(). + * Added stasis_app_control_mark_failed(). + * Added stasis_app_control_is_failed(). + * Fixed res_stasis_device_state so unsubscribe all works properly. + * Modified stasis_app_unregister() to unsubscribe from all event sources. + * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() + returns true.

+

http: + * Added ast_http_create_basic_auth_header().

+

md5: + * Added define for MD5_DIGEST_LENGTH.

+

tcptls: + * Added flag to ast_tcptls_session_args to suppress connection log messages + to give callers more control over logging.

+

http_websocket: + * Add flag to ast_websocket_client_options to suppress connection log messages + to give callers more control over logging. + * Added username and password to ast_websocket_client_options to support + outbound basic authentication. + * Added ast_websocket_result_to_str().

+

contrib: Add systemd service and timer files for malloc trim.

+

Author: Ben Ford + Date: 2025-04-16

+

Adds two files to the contrib/systemd/ directory that can be installed + to periodically run "malloc trim" on Asterisk. These files do nothing + unless they are explicitly moved to the correct location on the system. + Users who are experiencing Asterisk memory issues can use this service + to potentially help combat the problem. These files can also be + configured to change the start time and interval. See systemd.timer(5) + and systemd.time(7) for more information.

+

UserNote: Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system.

+

action_redirect: remove after_bridge_goto_info

+

Author: Peter Jannesen + Date: 2025-03-13

+

Under certain circumstances the context/extens/prio are stored in the + after_bridge_goto_info. This info is used when the bridge is broken by + for hangup of the other party. In the situation that the bridge is + broken by an AMI Redirect this info is not used but also not removed. + With the result that when the channel is put back in a bridge and the + bridge is broken the execution continues at the wrong + context/extens/prio.

+

Resolves: #1144

+

channel: Always provide cause code in ChannelHangupRequest.

+

Author: Joshua C. Colp + Date: 2025-04-16

+

When queueing a channel to be hung up a cause code can be + specified in one of two ways:

+
    +
  1. +

    ast_queue_hangup_with_cause + This function takes in a cause code and queues it as part + of the hangup request, which ultimately results in it being + set on the channel.

    +
    2. +
  2. +

    ast_channel_hangupcause_set + ast_queue_hangup + This combination sets the hangup cause on the channel before + queueing the hangup instead of as part of that process.

    +
    3. +
+

In the #2 case the ChannelHangupRequest event would not contain + the cause code. For consistency if a cause code has been set + on the channel it will now be added to the event.

+

Resolves: #1197

+

Add log-caller-id-name option to log Caller ID Name in queue log

+

Author: phoneben + Date: 2025-02-28

+

Add log-caller-id-name option to log Caller ID Name in queue log

+

This patch introduces a new global configuration option, log-caller-id-name, + to queues.conf to control whether the Caller ID name is logged when a call enters a queue.

+

When log-caller-id-name=yes, the Caller ID name is logged + as parameter 4 in the queue log, provided it’s allowed by the + existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), + the Caller ID name is omitted from the logs.

+

Fixes: #1091

+

UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted.

+

asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.

+

Author: George Joseph + Date: 2025-04-10

+

Commands in the "[startup_commands]" section of cli.conf have historically run + after all core and module initialization has been completed and just before + "Asterisk Ready" is printed on the console. This meant that if you + wanted to debug initialization of a specific module, your only option + was to turn on debug for everything by setting "debug" in asterisk.conf.

+

This commit introduces options to allow you to run CLI commands earlier in + the asterisk startup process.

+

A command with a value of "pre-init" will run just after logger initialization + but before most core, and all module, initialization.

+

A command with a value of "pre-module" will run just after all core + initialization but before all module initialization.

+

A command with a value of "fully-booted" (or "yes" for backwards + compatibility) will run as they always have been...after all + initialization and just before "Asterisk Ready" is printed on the console.

+

This means you could do this...

+

[startup_commands] + core set debug 3 res_pjsip.so = pre-module + core set debug 0 res_pjsip.so = fully-booted

+

This would turn debugging on for res_pjsip.so to catch any module + initialization debug messages then turn it off again after the module is + loaded.

+

UserNote: In cli.conf, you can now define startup commands that run before + core initialization and before module initialization.

+

app_confbridge: Prevent crash when publishing channel-less event.

+

Author: Sean Bright + Date: 2025-04-07

+

Resolves: #1190

+

ari_websockets: Fix frack if ARI config fails to load.

+

Author: George Joseph + Date: 2025-04-02

+

ari_ws_session_registry_dtor() wasn't checking that the container was valid + before running ao2_callback on it to shutdown registered sessions.

+

ARI: REST over Websocket

+

Author: George Joseph + Date: 2025-03-12

+

This commit adds the ability to make ARI REST requests over the same + websocket used to receive events.

+

For full details on how to use the new capability, visit...

+

https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/

+

Changes:

+ +

UpgradeNote: This commit adds the ability to make ARI REST requests over the same + websocket used to receive events. + See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/

+

audiohook.c: Add ability to adjust volume with float

+

Author: mkmer + Date: 2025-03-18

+

Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41.

+

This is accomplished by the following: + Convert internal variables to type float. + Always use ast_frame_adjust_volume_float() for adjustments. + Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). + Cast float to int in ast_audiohook_volume_get() + Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float.

+

This update maintains 100% backward compatibility.

+

Resolves: #1171

+

audiosocket: added support for DTMF frames

+

Author: Florent CHAUVEAU + Date: 2025-02-28

+

Updated the AudioSocket protocol to allow sending DTMF frames. + AST_FRAME_DTMF frames are now forwarded to the server, in addition to + AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF + with value 0x03 was added to the protocol. The payload is a 1-byte + ascii representing the DTMF digit (0-9,*,#...).

+

UserNote: The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...).

+

asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'

+

Author: Norm Harrison + Date: 2023-04-03

+

Co-authored-by: Florent CHAUVEAU florentch@pm.me

+

audiosocket: fix timeout, fix dialplan app exit, server address in logs

+

Author: Norm Harrison + Date: 2023-04-03

+ +

Co-authored-by: Florent CHAUVEAU florentch@pm.me

+

chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip show contact'

+

Author: Mark Murawski + Date: 2025-03-23

+

CLI 'pjsip show contact' does not show enough information. + One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact + This feature adds the same details as PJSIPShowContacts to the CLI

+

Resolves: #643

+

Update config.guess and config.sub

+

Author: Zhai Liangliang + Date: 2025-03-26

+

chan_pjsip: set correct Endpoint Device State on multiple channels

+

Author: Alexei Gradinari + Date: 2025-03-25

+
    +
  1. +

    When one channel is placed on hold, the device state is set to ONHOLD + without checking other channels states. + In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN + to calculate aggregate device state of all active channels.

    +
    2. +
  2. +

    The current implementation incorrectly classifies channels in use. + The only channels that has the states: UP, RING and BUSY are considered as "in use". + A channel should be considered "in use" if its state is anything other than + DOWN or RESERVED.

    +
    3. +
  3. +

    Currently, if the number of channels "in use" is greater than device_state_busy_at, + the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate + device state. + The endpoint device state should be BUSY if the number of channels "in use" is greater + than or equal to device_state_busy_at.

    +
    4. +
+

Fixes: #1181

+

file.c: missing "custom" sound files should not generate warning logs

+

Author: Allan Nathanson + Date: 2025-03-18

+

With sounds_search_custom_dir = yes we first look to see if a sound file + is present in the "custom" sound directory before looking in the standard + sound directories. We should not be issuing a WARNING log message if a + sound cannot be found in the "custom" directory.

+

Resolves: https://github.com/asterisk/asterisk/issues/1170

+

documentation: Update Gosub, Goto, and add new documentationtype.

+

Author: Ben Ford + Date: 2025-03-14

+

Gosub and Goto were not displaying their syntax correctly on the docs + site. This change adds a new way to specify an optional context, an + optional extension, and a required priority that the xml stylesheet can + parse without having to know which optional parameters come in which + order. In Asterisk, it looks like this:

+
parameter name="context" documentationtype="dialplan_context"
+parameter name="extension" documentationtype="dialplan_extension"
+parameter name="priority" documentationtype="dialplan_priority" required="true"
+
+

The stylesheet will ignore the context and extension parameters, but for + priority, it will automatically inject the following:

+
[[context,]extension,]priority
+
+

This is the correct oder for applications such as Gosub and Goto.

+

res_config_curl.c: Remove unnecessary warnings.

+

Author: Sean Bright + Date: 2025-03-17

+

Resolves: #1164

+

README.md: Updates and Fixes

+

Author: George Joseph + Date: 2025-03-05

+ +

Going forward, the release process will create HTML versions of the README + and change log and will update the link in the README to the current + change log for the branch...

+ +

Resolves: #1131

+

res_rtp_asterisk.c: Don't truncate spec-compliant ice-ufrag or ice-pwd.

+

Author: Sean Bright + Date: 2025-03-07

+

RFC 8839[1] indicates that the ice-ufrag and ice-pwd attributes + can be up to 256 bytes long. While we don't generate values of that + size, we should be able to accomodate them without truncating.

+
    +
  1. https://www.rfc-editor.org/rfc/rfc8839#name-ice-ufrag-and-ice-pwd-attri
    2. +
+

fix: Correct default flag for tcp_keepalive_enable option

+

Author: Joshua Elson + Date: 2025-03-06

+

Resolves an issue where the tcp_keepalive_enable option was not properly enabled in the sample configuration due to an incorrect default flag setting.

+

Fixes: #1149

+

docs: AMI documentation fixes.

+

Author: Sean Bright + Date: 2025-02-18

+

Most of this patch is adding missing PJSIP-related event + documentation, but the one functional change was adding a sorcery + to-string handler for endpoint's redirect_method which was not + showing up in the AMI event details or pjsip show endpoint + <endpoint> output.

+

The rest of the changes are summarized below:

+ +

config.c: #include of non-existent file should not crash

+

Author: Allan Nathanson + Date: 2025-03-03

+

Corrects a segmentation fault when a configuration file has a #include + statement that referenced a file that does not exist.

+

Resolves: https://github.com/asterisk/asterisk/issues/1139

+

manager.c: Check for restricted file in action_createconfig.

+

Author: George Joseph + Date: 2025-03-03

+

The CreateConfig manager action now ensures that a config file can + only be created in the AST_CONFIG_DIR unless live_dangerously is set.

+

Resolves: #1122

+

swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().

+

Author: George Joseph + Date: 2025-03-04

+

Recent python versions complain when backslashes in strings create invalid + escape sequences. This causes issues for strings used as regex patterns like + '^List\[(.*)\]$' where you want the regex parser to treat [ and ] + as literals. Double-backslashing is one way to fix it but simply converting + the string to a raw string re.match(r'^List\[(.*)\]$', text) is easier + and less error prone.

+

res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer.

+

Author: Sean Bright + Date: 2025-02-24

+

Found while reviewing #1128

+

docs: Fix typos in cdr/ Found via codespell

+

Author: Luz Paz + Date: 2025-02-12

+

docs: Fix various typos in channels/ Found via codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues

+

Author: Luz Paz + Date: 2025-02-04

+

docs: Fix various typos in main/ Found via codespell -q 3 -S "./CREDITS" -L abd,asent,atleast,childrens,contentn,crypted,dne,durationm,exten,inout,leapyear,nd,oclock,offsetp,ot,parm,parms,requestor,ser,slanguage,slin,thirdparty,varn,varns,ues

+

Author: Luz Paz + Date: 2025-02-04

+

bridging: Fix multiple bridging issues causing SEGVs and FRACKs.

+

Author: George Joseph + Date: 2025-01-22

+

Issues:

+ +

Changes:

+ +

Resolves: #211

+

bridge_channel: don't set cause code on channel during bridge delete if already set

+

Author: Mike Bradeen + Date: 2025-02-18

+

Due to a potential race condition via ARI when hanging up a channel hangup with cause + while also deleting a bridge containing that channel, the bridge delete can over-write + the hangup cause code resulting in Normal Call Clearing instead of the set value.

+

With this change, bridge deletion will only set the hangup code if it hasn't been + previously set.

+

Resolves: #1124

+

res_config_pgsql: Fix regression that removed dbname config.

+

Author: George Joseph + Date: 2025-02-11

+

A recent commit accidentally removed the code that sets dbname. + This commit adds it back in.

+

Resolves: #1119

+

res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.

+

Author: George Joseph + Date: 2025-02-05

+

The verification check for missing or anonymous callerid was happening before + the endpoint's profile was retrieved which meant that the failure_action + parameter wasn't available. Therefore, if verification was enabled and there + was no callerid or it was "anonymous", the call was immediately terminated + instead of giving the dialplan the ability to decide what to do with the call.

+ +

Resolves: #1112

+

resource_channels.c: Fix memory leak in ast_ari_channels_external_media.

+

Author: George Joseph + Date: 2025-02-04

+

Between ast_ari_channels_external_media(), external_media_rtp_udp(), + and external_media_audiosocket_tcp(), the variables structure being passed + around wasn't being cleaned up properly when there was a failure.

+ +

Resolves: #1109

+

ari/pjsip: Make it possible to control transfers through ARI

+

Author: Holger Hans Peter Freyther + Date: 2024-06-15

+

Introduce a ChannelTransfer event and the ability to notify progress to + ARI. Implement emitting this event from the PJSIP channel instead of + handling the transfer in Asterisk when configured.

+

Introduce a dialplan function to the PJSIP channel to switch between the + "core" and "ari-only" behavior.

+

UserNote: Call transfers on the PJSIP channel can now be controlled by + ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) + dialplan function.

+

channel.c: Remove dead AST_GENERATOR_FD code.

+

Author: Sean Bright + Date: 2025-02-06

+

Nothing ever sets the AST_GENERATOR_FD, so this block of code will + never execute. It also is the only place where the generate callback + is called with the channel lock held which made it difficult to reason + about the thread safety of ast_generators.

+

In passing, also note that AST_AGENT_FD isn't used either.

+

func_strings.c: Prevent SEGV in HASH single-argument mode.

+

Author: George Joseph + Date: 2025-01-30

+

When in single-argument mode (very rarely used), a malformation of a column + name (also very rare) could cause a NULL to be returned when retrieving the + channel variable for that column. Passing that to strncat causes a SEGV. We + now check for the NULL and print a warning message.

+

Resolves: #1101

+

docs: Add version information to AGI command XML elements.

+

Author: George Joseph + Date: 2025-01-24

+

This process was a bit different than the others because everything + is in the same file, there's an array that contains the command + names and their handler functions, and the last command was created + over 15 years ago.

+ +

docs: Fix minor typo in MixMonitor AMI action

+

Author: Jeremy Lainé + Date: 2025-01-28

+

The Options argument was erroneously documented as lowercase + options.

+

utils: Disable old style definition warnings for libdb.

+

Author: Naveen Albert + Date: 2025-01-23

+

Newer versions of gcc now warn about old style definitions, such + as those in libdb, which causes compilation failure with DEVMODE + enabled. Ignore these warnings for libdb.

+

Resolves: #1085

+

rtp.conf.sample: Correct stunaddr example.

+

Author: fabriziopicconi + Date: 2024-09-25

+

docs: Add version information to ARI resources and methods.

+

Author: George Joseph + Date: 2025-01-27

+ +

docs: Indent tags.

+

Author: Sean Bright + Date: 2025-01-23

+

Also updates the 'since' of applications/functions that existed before + XML documentation was introduced (1.6.2.0).

+

res_pjsip_authenticator_digest: Make correct error messages appear again.

+

Author: George Joseph + Date: 2025-01-28

+

When an incoming request can't be matched to an endpoint, the "artificial" + auth object is used to create a challenge to return in a 401 response and we + emit a "No matching endpoint found" log message. If the client then responds + with an Authorization header but the request still can't be matched to an + endpoint, the verification will fail and, as before, we'll create a challenge + to return in a 401 response and we emit a "No matching endpoint found" log + message. HOWEVER, because there WAS an Authorization header and it failed + verification, we should have also been emitting a "Failed to authenticate" + log message but weren't because there was a check that short-circuited that + it if the artificial auth was used. Since many admins use the "Failed to + authenticate" message with log parsers like fail2ban, those attempts were not + being recognized as suspicious.

+

Changes:

+ +

Resolves: #1095

+

alembic: Database updates required.

+

Author: George Joseph + Date: 2025-01-28

+

This commit doesn't actually change anything. It just adds the following + upgrade notes that were omitted from the original commits.

+

Resolves: #1097

+

UpgradeNote: Two commits in this release... + 'Add SHA-256 and SHA-512-256 as authentication digest algorithms' + 'res_pjsip: Add new AOR option "qualify_2xx_only"' + ...have modified alembic scripts for the following database tables: ps_aors, + ps_contacts, ps_auths, ps_globals. If you don't use the scripts to update + your database, reads from those tables will succeeed but inserts into the + ps_contacts table by res_pjsip_registrar will fail.

+

res_pjsip: Fix startup/reload memory leak in config_auth.

+

Author: George Joseph + Date: 2025-01-23

+

An issue in config_auth.c:ast_sip_auth_digest_algorithms_vector_init() was + causing double allocations for the two supported_algorithms vectors to the + tune of 915 bytes. The leak only happens on startup and when a reload is done + and doesn't get bigger with the number of auth objects defined.

+ +

docs: Add version information to application and function XML elements

+

Author: George Joseph + Date: 2025-01-23

+ +

The applications and functions defined in the following files were done by + hand because the XML was extracted from the C source file relatively recently. + * channels/pjsip/dialplan_functions_doc.xml + * main/logger_doc.xml + * main/manager_doc.xml + * res/res_geolocation/geoloc_doc.xml + * res/res_stir_shaken/stir_shaken_doc.xml

+

docs: Add version information to manager event instance XML elements

+

Author: George Joseph + Date: 2025-01-20

+ +

The events defined in res/res_pjsip/pjsip_manager.xml were done by hand + because the XML was extracted from the C source file relatively recently.

+

Two bugs were fixed along the way...

+ +

LICENSE: Update company name, email, and address.

+

Author: Joshua C. Colp + Date: 2025-01-21

+

res_prometheus.c: Set Content-Type header on /metrics response.

+

Author: Sean Bright + Date: 2025-01-21

+

This should resolve the Prometheus error:

+
+

Error scraping target: non-compliant scrape target + sending blank Content-Type and no + fallback_scrape_protocol specified for target.

+
+

Resolves: #1075

+

README.md, asterisk.c: Update Copyright Dates

+

Author: George Joseph + Date: 2025-01-20

+

docs: Add version information to configObject and configOption XML elements

+

Author: George Joseph + Date: 2025-01-16

+

Most of the configObjects and configOptions that are implemented with + ACO or Sorcery now have <since>/<version> elements added. There are + probably some that the script I used didn't catch. The version tags were + determined by the following... + * Do a git blame on the API call that created the object or option. + * From the commit hash, grab the summary line. + * Do a git log --grep <summary> to find the cherry-pick commits in all + branches that match. + * Do a git patch-id to ensure the commits are all related and didn't get + a false match on the summary. + * Do a git tag --contains <commit> to find the tags that contain each + commit. + * Weed out all tags not ..0. + * Sort and discard any .0.0 and following tags where the commit + appeared in an earlier branch. + * The result is a single tag for each branch where the API was last touched.

+

configObjects and configOptions elements implemented with the base + ast_config APIs were just not possible to find due to the non-deterministic + way they are accessed.

+

Also note that if the API call was on modified after it was added, the + version will be the one it was last modified in.

+

Final note: The configObject and configOption elements were introduced in + 12.0.0 so options created before then may not have any XML documentation.

+

res_pjsip_authenticator_digest: Fix issue with missing auth and DONT_OPTIMIZE

+

Author: George Joseph + Date: 2025-01-17

+

The return code fom digest_check_auth wasn't explicitly being initialized. + The return code also wasn't explicitly set to CHALLENGE when challenges + were sent. When optimization was turned off (DONT_OPTIMIZE), the compiler + was setting it to "0"(CHALLENGE) which worked fine. However, with + optimization turned on, it was setting it to "1" (SUCCESS) so if there was + no incoming Authorization header, the function was returning SUCCESS to the + distributor allowing the request to incorrectly succeed.

+

The return code is now initialized correctly and is now explicitly set + to CHALLENGE when we send challenges.

+

ast_tls_cert: Add option to skip passphrase for CA private key.

+

Author: Naveen Albert + Date: 2025-01-14

+

Currently, the ast_tls_cert file is hardcoded to use the -des3 option + for 3DES encryption, and the script needs to be manually modified + to not require a passphrase. Add an option (-e) that disables + encryption of the CA private key so no passphrase is required.

+

Resolves: #1064

+

chan_iax2: Avoid unnecessarily backlogging non-voice frames.

+

Author: Naveen Albert + Date: 2025-01-09

+

Currently, when receiving an unauthenticated call, we keep track + of the negotiated format in the chosenformat, which allows us + to later create the channel using the right format. However, + this was not done for authenticated calls. This meant that in + certain circumstances, if we had not yet received a voice frame + from the peer, only certain other types of frames (e.g. text), + there were no variables containing the appropriate frame. + This led to problems in the jitterbuffer callback where we + unnecessarily bailed out of retrieving a frame from the jitterbuffer. + This was logic intentionally added in commit 73103bdcd5b342ce5dfa32039333ffadad551151 + in response to an earlier regression, and while this prevents + crashes, it also backlogs legitimate frames unnecessarily.

+

The abort logic was initially added because at this point in the + code, we did not have the negotiated format available to us. + However, it should always be available to us as a last resort + in chosenformat, so we now pull it from there if needed. This + allows us to process frames the jitterbuffer even if voicefmt + and peerfmt aren't set and still avoid the crash. The failsafe + logic is retained, but now it shouldn't be triggered anymore.

+

Resolves: #1054

+

config.c: fix #tryinclude being converted to #include on rewrite

+

Author: Allan Nathanson + Date: 2024-09-16

+

Correct an issue in ast_config_text_file_save2() when updating configuration + files with "#tryinclude" statements. The API currently replaces "#tryinclude" + with "#include". The API also creates empty template files if the referenced + files do not exist. This change resolves these problems.

+

Resolves: https://github.com/asterisk/asterisk/issues/920

+

sig_analog: Add Last Number Redial feature.

+

Author: Naveen Albert + Date: 2023-11-10

+

This adds the Last Number Redial feature to + simple switch.

+

UserNote: Users can now redial the last number + called if the lastnumredial setting is set to yes.

+

Resolves: #437

+

docs: Various XML fixes

+

Author: George Joseph + Date: 2025-01-15

+ +

strings.c: Improve numeric detection in ast_strings_match().

+

Author: Sean Bright + Date: 2025-01-15

+

Essentially, we were treating 1234x1234 and 1234x5678 as 'equal' + because we were able to convert the prefix of each of these strings to + the same number.

+

Resolves: #1028

+

docs: Enable since/version handling for XML, CLI and ARI documentation

+

Author: George Joseph + Date: 2025-01-09

+ +

NOTE: No versioning information is actually added in this commit. + Those will be added separately and instructions for adding and maintaining + them will be published on the documentation site at a later date.

+

logger.h: Fix build when AST_DEVMODE is not defined.

+

Author: Artem Umerov + Date: 2025-01-13

+

Resolves: #1058

+

dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's media argument.

+

Author: Sean Bright + Date: 2025-01-14

+

Resolves: #1023

+

samples: Use "asterisk" instead of "postgres" for username

+

Author: Abdelkader Boudih + Date: 2025-01-07

+

manager: Add <since> tags for all AMI actions.

+

Author: Sean Bright + Date: 2025-01-02

+

logger.c fix: malformed JSON template

+

Author: Steffen Arntz + Date: 2025-01-08

+

this typo was mentioned before, but never got fixed. + https://community.asterisk.org/t/logger-cannot-log-long-json-lines-properly/87618/6

+

manager.c: Rename restrictedFile to is_restricted_file.

+

Author: Sean Bright + Date: 2025-01-09

+

Also correct the spelling of 'privileges.'

+

res_config_pgsql: normalize database connection option with cel and cdr by supporting new options name

+

Author: Abdelkader Boudih + Date: 2025-01-08

+

res_pjproject: Fix typo (OpenmSSL->OpenSSL)

+

Author: Stanislav Abramenkov + Date: 2025-01-10

+

Fix typo (OpenmSSL->OpenSSL) mentioned by bkford in #972

+

Add SHA-256 and SHA-512-256 as authentication digest algorithms

+

Author: George Joseph + Date: 2024-10-17

+ +

The new algorithms are only available with the bundled version + of pjproject, or an external version > 2.14.1. OpenSSL version + 1.1.1 or greater is required to support SHA-512-256.

+

Resolves: #948

+

UserNote: The SHA-256 and SHA-512-256 algorithms are now available + for authentication as both a UAS and a UAC.

+

config.c: retain leading whitespace before comments

+

Author: Allan Nathanson + Date: 2024-10-30

+

Configurations loaded with the ast_config_load2() API and later written + out with ast_config_text_file_save2() will have any leading whitespace + stripped away. The APIs should make reasonable efforts to maintain the + content and formatting of the configuration files.

+

This change retains any leading whitespace from comment lines that start + with a ";".

+

Resolves: https://github.com/asterisk/asterisk/issues/970

+

config.c: Fix off-nominal reference leak.

+

Author: Sean Bright + Date: 2025-01-07

+

This was identified and fixed by @Allan-N in #918 but it is an + important fix in its own right.

+

The fix here is slightly different than Allan's in that we just move + the initialization of the problematic AO2 container to where it is + first used.

+

Fixes #1046

+

normalize contrib/ast-db-manage/queue_log.ini.sample

+

Author: Abdelkader Boudih + Date: 2025-01-05

+

Add C++ Standard detection to configure and fix a new C++20 compile issue

+

Author: George Joseph + Date: 2025-01-03

+ +

chan_dahdi: Fix wrong channel state when RINGING recieved.

+

Author: Naveen Albert + Date: 2024-12-16

+

Previously, when AST_CONTROL_RINGING was received by + a DAHDI device, it would set its channel state to + AST_STATE_RINGING. However, an analysis of the codebase + and other channel drivers reveals RINGING corresponds to + physical power ringing, whereas AST_STATE_RING should be + used for audible ringback on the channel. This also ensures + the correct device state is returned by the channel state + to device state conversion.

+

Since there seems to be confusion in various places regarding + AST_STATE_RING vs. AST_STATE_RINGING, some documentation has + been added or corrected to clarify the actual purposes of these + two channel states, and the associated device state mapping.

+

An edge case that prompted this fix, but isn't explicitly + addressed here, is that of an incoming call to an FXO port. + The channel state will be "Ring", which maps to a device state + of "In Use", not "Ringing" as would be more intuitive. However, + this is semantic, since technically, Asterisk is treating this + the same as any other incoming call, and so "Ring" is the + semantic state (put another way, Asterisk isn't ringing anything, + like in the cases where channels are in the "Ringing" state).

+

Since FXO ports don't currently support Call Waiting, a suitable + workaround for the above would be to ignore the device state and + instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state))) + since it will be Ring if the FXO port is idle (but a call is ringing + on it) and Up if the FXO port is actually in use. (In both cases, + the device state would misleadingly be "In Use".)

+

Resolves: #1029

+

Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016

+

Author: Stanislav Abramenkov + Date: 2024-12-03

+

UserNote: Bundled pjproject has been upgraded to 2.15.1. For more + information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.15.1

+

gcc14: Fix issues caught by gcc 14

+

Author: George Joseph + Date: 2025-01-03

+ +

Header fixes for compiling C++ source files

+

Author: George Joseph + Date: 2024-12-31

+

A few tweaks needed to be done to some existing header files to allow them to + be compiled when included from C++ source files.

+

logger.h had declarations for ast_register_verbose() and + ast_unregister_verbose() which caused C++ issues but those functions were + actually removed from logger.c many years ago so the declarations were just + removed from logger.h.

+

Add ability to pass arguments to unit tests from the CLI

+

Author: George Joseph + Date: 2024-12-27

+

Unit tests can now be passed custom arguments from the command + line. For example, the following command would run the "mytest" test + in the "/main/mycat" category with the option "myoption=54"

+

CLI> test execute category /main/mycat name mytest options myoption=54

+

You can also pass options to an entire category...

+

CLI> test execute category /main/mycat options myoption=54

+

Basically, everything after the "options" keyword is passed verbatim to + the test which must decide what to do with it.

+ +

res_pjsip: Add new AOR option "qualify_2xx_only"

+

Author: Kent + Date: 2024-12-03

+

Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify + feature to mark a contact as available only if an OPTIONS request + returns a 2XX response. If the option is not specified or is false, + any response to the OPTIONS request marks the contact as available.

+

UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only" + option that can be set so that only 2XX responses to OPTIONS requests + used to qualify a contact will mark the contact as available.

+

res_odbc: release threads from potential starvation.

+

Author: Jaco Kroon + Date: 2024-12-10

+

Whenever a slot is freed up due to a failed connection, wake up a waiter + before failing.

+

In the case of a dead connection there could be waiters, for example, + let's say two threads tries to acquire objects at the same time, with + one in the cached connections, one will acquire the dead connection, and + the other will enter into the wait state. The thread with the dead + connection will clear up the dead connection, and then attempt a + re-acquire (at this point there cannot be cached connections else the + other thread would have received that and tried to clean up), as such, + at this point we're guaranteed that either there are no waiting threads, + or that the maxconnections - connection_cnt threads will attempt to + re-acquire connections, and then either succeed, using those + connections, or failing, and then signalling to release more waiters.

+

Also fix the pointer log for ODBC handle %p dead which would always + reflect NULL.

+

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

+

app_queue: indicate the paused state of a dynamically added member in queue_log.

+

Author: Sperl Viktor + Date: 2024-12-05

+

Fixes: #1021

+

Allow C++ source files (as extension .cc) in the main directory

+

Author: George Joseph + Date: 2024-12-09

+

Although C++ files (as extension .cc) have been handled in the module + directories for many years, the main directory was missing one line in its + Makefile that prevented C++ files from being recognised there.

+

format_gsm.c: Added mime type

+

Author: Alexey Khabulyak + Date: 2024-12-03

+

Sometimes it's impossible to get a file extension from URL + (eg. http://example.com/gsm/your) so we have to rely on content-type header. + Currenly, asterisk does not support content-type for gsm format(unlike wav). + Added audio/gsm according to https://www.rfc-editor.org/rfc/rfc4856.html

+

func_uuid: Add a new dialplan function to generate UUIDs

+

Author: Maksim Nesterov + Date: 2024-12-01

+

This function is useful for uniquely identifying calls, recordings, and other entities in distributed environments, as well as for generating an argument for the AudioSocket application.

+

app_queue: allow dynamically adding a queue member in paused state.

+

Author: Sperl Viktor + Date: 2024-11-27

+

Fixes: #1007

+

UserNote: use the p option of AddQueueMember() for paused member state. + Optionally, use the r(reason) option to specify a custom reason for the pause.

+

chan_iax2: Add log message for rejected calls.

+

Author: Naveen Albert + Date: 2023-11-06

+

Add a log message for a path that currently silently drops IAX2 + frames without indicating that anything is wrong.

+

chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams

+

Author: Maximilian Fridrich + Date: 2024-12-02

+

Currently, when a chan_pjsip channel receives a VIDUPDATE indication, + an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is + enabled on that endpoint. This restriction does not really make sense.

+

Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as + is the case with VP8, VP9, and H.265 streams).

+

Resolves: #1013

+

audiohook.c: resolving the issue with audiohook both reading when packet loss on one side of the call

+

Author: Tinet-mucw + Date: 2024-08-22

+

When there is 0% packet loss on one side of the call and 15% packet loss on the other side, reading frame is often failed when reading direction_both audiohook. when read_factory available = 0, write_factory available = 320; i think write factory is usable read; because after reading one frame, there is still another frame that can be read together with the next read factory frame.

+

Resolves: #851

+

res_curl.conf.sample: clean up sample configuration and add new SSL options

+

Author: Mike Pultz + Date: 2024-11-21

+

This update properly documents all the current configuration options supported + by the curl implementation, including the new ssl_* options.

+

res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF

+

Author: Alexey Vasilyev + Date: 2024-11-25

+

Fixes #1004

+

manager.c: Add Processed Call Count to CoreStatus output

+

Author: Mike Pultz + Date: 2024-11-21

+

This update adds the processed call count to the CoreStatus AMI Action responsie. This output is + similar to the values returned by "core show channels" or "core show calls" in the CLI.

+

UserNote: The current processed call count is now returned as CoreProcessedCalls from the + CoreStatus AMI Action.

+

func_curl.c: Add additional CURL options for SSL requests

+

Author: Mike Pultz + Date: 2024-11-09

+

This patch adds additional CURL TLS options / options to support mTLS authenticated requests:

+ +

UserNote: The following new configuration options are now available + in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost' + (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath' + (CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype' + (CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype', + (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the + libcurl documentation for more details.

+

sig_analog: Fix regression with FGD and E911 signaling.

+

Author: Naveen Albert + Date: 2024-11-14

+

Commit 466eb4a52b69e6dead7ebba13a83f14ef8a559c1 introduced a regression + which completely broke Feature Group D and E911 signaling, by removing + the call to analog_my_getsigstr, which affected multiple switch cases. + Restore the original behavior for all protocols except Feature Group C + CAMA (MF), which is all that patch was attempting to target.

+

Resolves: #993

+

main/stasis_channels.c: Fix crash when setting a global variable with invalid UTF8 characters

+

Author: James Terhune + Date: 2024-11-18

+

Add check for null value of chan before referencing it with ast_channel_name()

+

Resolves: #999

+

res_stir_shaken: Allow sending Identity headers for unknown TNs

+

Author: George Joseph + Date: 2024-11-08

+

Added a new option "unknown_tn_attest_level" to allow Identity + headers to be sent when a callerid TN isn't explicitly configured + in stir_shaken.conf. Since there's no TN object, a private_key_file + and public_cert_url must be configured in the attestation or profile + objects.

+

Since "unknown_tn_attest_level" uses the same enum as attest_level, + some of the sorcery macros had to be refactored to allow sharing + the enum and to/from string conversion functions.

+

Also fixed a memory leak in crypto_utils:pem_file_cb().

+

Resolves: #921

+

UserNote: You can now set the "unknown_tn_attest_level" option + in the attestation and/or profile objects in stir_shaken.conf to + enable sending Identity headers for callerid TNs not explicitly + configured.

+

manager.c: Restrict ListCategories to the configuration directory.

+

Author: Ben Ford + Date: 2024-12-17

+

When using the ListCategories AMI action, it was possible to traverse + upwards through the directories to files outside of the configured + configuration directory. This action is now restricted to the configured + directory and an error will now be returned if the specified file is + outside of this limitation.

+

Resolves: #GHSA-33x6-fj46-6rfh

+

UserNote: The ListCategories AMI action now restricts files to the + configured configuration directory.

+

res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T

+

Author: George Joseph + Date: 2024-11-15

+

The suppress_moh_on_sendonly endpoint option should have been + defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES + in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.

+

Also updated contrib/ast-db-manage/README.md to indicate that + AST_BOOL_VALUES should always be used and provided an example.

+

Resolves: #995

+

res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"

+

Author: George Joseph + Date: 2024-11-05

+

Normally, when one party in a call sends Asterisk an SDP with + a "sendonly" or "inactive" attribute it means "hold" and causes + Asterisk to start playing MOH back to the other party. This can be + problematic if it happens at certain times, such as in a 183 + Progress message, because the MOH will replace any early media you + may be playing to the calling party. If you set this option + to "yes" on an endpoint and the endpoint receives an SDP + with "sendonly" or "inactive", Asterisk will NOT play MOH back to + the other party.

+

Resolves: #979

+

UserNote: The new "suppress_moh_on_sendonly" endpoint option + can be used to prevent playing MOH back to a caller if the remote + end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.

+

res_pjsip.c: Fix Contact header rendering for IPv6 addresses.

+

Author: Sean Bright + Date: 2024-11-08

+

Fix suggested by @nvsystems.

+

Fixes #985

+

samples: remove and/or change some wiki mentions

+

Author: chrsmj + Date: 2024-11-01

+

Cleaned some dead links. Replaced word wiki with + either docs or link to https://docs.asterisk.org/

+

Resolves: #974

+

func_pjsip_aor/contact: Fix documentation for contact ID

+

Author: George Joseph + Date: 2024-11-09

+

Clarified the use of the contact ID returned from PJSIP_AOR.

+

Resolves: #990

+

res_pjsip: Move tenantid to end of ast_sip_endpoint

+

Author: George Joseph + Date: 2024-11-06

+

The tenantid field was originally added to the ast_sip_endpoint + structure at the end of the AST_DECLARE_STRING_FIELDS block. This + caused everything after it in the structure to move down in memory + and break ABI compatibility. It's now at the end of the structure + as an AST_STRING_FIELD_EXTENDED. Given the number of string fields + in the structure now, the initial string field allocation was + also increased from 64 to 128 bytes.

+

Resolves: #982

+

pjsip_transport_events: handle multiple addresses for a domain

+

Author: Thomas Guebels + Date: 2024-10-29

+

The key used for transport monitors was the remote host name for the + transport and not the remote address resolved for this domain.

+

This was problematic for domains returning multiple addresses as several + transport monitors were created with the same key.

+

Whenever a subsystem wanted to register a callback it would always end + up attached to the first transport monitor with a matching key.

+

The key used for transport monitors is now the remote address and port + the transport actually connected to.

+

Fixes: #932

+

func_evalexten: Add EVAL_SUB function.

+

Author: Naveen Albert + Date: 2024-10-17

+

This adds an EVAL_SUB function, which is similar to the existing + EVAL_EXTEN function but significantly more powerful, as it allows + executing arbitrary dialplan and capturing its return value as + the function's output. While EVAL_EXTEN should be preferred if it + is possible to use it, EVAL_SUB can be used in a wider variety + of cases and allows arbitrary computation to be performed in + a dialplan function call, leveraging the dialplan.

+

Resolves: #951

+

res_srtp: Change Unsupported crypto suite msg from verbose to debug

+

Author: George Joseph + Date: 2024-11-01

+

There's really no point in spamming logs with a verbose message + for every unsupported crypto suite an older client may send + in an SDP. If none are supported, there will be an error or + warning.

+

Add res_pjsip_config_sangoma external module.

+

Author: Ben Ford + Date: 2024-11-01

+

Adds res_pjsip_config_sangoma as an external module that can be + downloaded via menuselect. It lives under the Resource Modules section.

+

app_mixmonitor: Add 'D' option for dual-channel audio.

+

Author: Ben Ford + Date: 2024-10-28

+

Adds the 'D' option to app_mixmonitor that interleaves the input and + output frames of the channel being recorded in the monitor output frame. + This allows for two streams in the recording: the transmitted audio and + the received audio. The 't' and 'r' options are compatible with this.

+

Fixes: #945

+

UserNote: The MixMonitor application now has a new 'D' option which + interleaves the recorded audio in the output frames. This allows for + stereo recording output with one channel being the transmitted audio and + the other being the received audio. The 't' and 't' options are + compatible with this.

+

pjsip_transport_events: Avoid monitor destruction

+

Author: Thomas Guebels + Date: 2024-10-28

+

When a transport is disconnected, several events can arrive following + each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it + will trigger the destruction of the transport monitor object. The lookup + for the transport monitor to destroy is done using the transport key, + that contains the transport destination host:port.

+

A reconnect attempt by pjsip will be triggered as soon something needs to + send a packet using that transport. This can happen directly after a + disconnect since ca

+

Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will + also try to trigger the destruction of the transport monitor if not + already done. Since the lookup for the transport monitor to destroy is + done using the transport key, it can match newly created transports + towards the same destination and destroy their monitor object.

+

Because of this, it was sometimes not possible to monitor a transport + after one or more disconnections.

+

This fix adds an additional check on the transport pointer to ensure + only a monitor for that specific transport is removed.

+

Fixes: #923

+

app_dial: Fix progress timeout calculation with no answer timeout.

+

Author: Naveen Albert + Date: 2024-10-16

+

If to_answer is -1, simply comparing to see if the progress timeout + is smaller than the answer timeout to prefer it will fail. Add + an additional check that chooses the progress timeout if there is + no answer timeout (or as before, if the progress timeout is smaller).

+

Resolves: #821

+

pjproject_bundled: Tweaks to support out-of-tree development

+

Author: George Joseph + Date: 2024-10-17

+ +

See third-party/pjproject/README-hacking.md for more info on building + pjproject "out-of-tree".

+

Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets."

+

Author: Sean Bright + Date: 2024-10-07

+

This reverts commit cb5e3445be6c55517c8d05aca601b648341f8ae9.

+

The original change from 16 to 15 bit sequence numbers was predicated + on the following from the now-defunct libSRTP FAQ on sourceforge.net:

+
+

Q6. The use of implicit synchronization via ROC seems +dangerous. Can senders and receivers lose ROC synchronization?

+

A. It is possible to lose ROC synchronization between sender and +receiver(s), though it is not likely in practice, and practical +steps can be taken to avoid it. A burst loss of 2^16 packets or more +will always break synchronization. For example, a conversational +voice codec that sends 50 packets per second will have its ROC +increment about every 22 minutes. A network with a burst of packet +loss that long has problems other than ROC synchronization.

+

There is a higher sensitivity to loss at the very outset of an SRTP +stream. If the sender's initial sequence number is close to the +maximum value of 2^16-1, and all packets are lost from the initial +packet until the sequence number cycles back to zero, the sender +will increment its ROC, but the receiver will not. The receiver +cannot determine that the initial packets were lost and that +sequence-number rollover has occurred. In this case, the receiver's +ROC would be zero whereas the sender's ROC would be one, while their +sequence numbers would be so close that the ROC-guessing algorithm +could not detect this fact.

+

There is a simple solution to this problem: the SRTP sender should +randomly select an initial sequence number that is always less than +2^15. This ensures correct SRTP operation so long as fewer than 2^15 +initial packets are lost in succession, which is within the maximum +tolerance of SRTP packet-index determination (see Appendix A and +page 14, first paragraph of RFC 3711). An SRTP receiver should +carefully implement the index-guessing algorithm. A naive +implementation can unintentionally guess the value of +0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15 +and the locally stored SEQ and ROC are zero. (This can happen when +the implementation fails to treat those zero values as a special +case.)

+

When ROC synchronization is lost, the receiver will not be able to +properly process the packets. If anti-replay protection is turned +on, then the desynchronization will appear as a burst of replay +check failures. Otherwise, if authentication is being checked, then +it will appear as a burst of authentication failures. Otherwise, if +encryption is being used, the desynchronization may not be detected +by the SRTP layer, and the packets may be improperly decrypted.

+
+

However, modern libSRTP (as of 1.0.1[1]) now mentions the following in + their README.md[2]:

+
+

The sequence number in the rtp packet is used as the low 16 bits of +the sender's local packet index. Note that RTP will start its +sequence number in a random place, and the SRTP layer just jumps +forward to that number at its first invocation. An earlier version +of this library used initial sequence numbers that are less than +32,768; this trick is no longer required as the +rdbx_estimate_index(...) function has been made smarter.

+
+

So truncating our initial sequence number to 15 bit is no longer + necessary.

+
    +
  1. https://github.com/cisco/libsrtp/blob/0eb007f0dc611f27cbfe0bf9855ed85182496cec/CHANGES#L271-L289
    2. +
  2. https://github.com/cisco/libsrtp/blob/2de20dd9e9c8afbaf02fcf5d4048ce1ec9ddc0ae/README.md#implementation-notes
    3. +
+

core_unreal.c: Fix memory leak in ast_unreal_new_channels()

+

Author: George Joseph + Date: 2024-10-15

+

When the channel tech is multistream capable, the reference to + chan_topology was passed to the new channel. When the channel tech + isn't multistream capable, the reference to chan_topology was never + released. "Local" channels are multistream capable so it didn't + affect them but the confbridge "CBAnn" and the bridge_media + "Recorder" channels are not so they caused a leak every time one + of them was created.

+

Also added tracing to ast_stream_topology_alloc() and + stream_topology_destroy() to assist with debugging.

+

Resolves: #938

+

dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin

+

Author: Allan Nathanson + Date: 2024-09-29

+

The dnsmgr_refresh() function checks to see if the IP address associated + with a name/service has changed. The gotcha is that the ast_get_ip_or_srv() + function only returns the first IP address returned by the DNS query. If + there are multiple IPs associated with the name and the returned order is + not consistent (e.g. with DNS round-robin) then the other IP addresses are + not included in the comparison and the entry is flagged as changed even + though the IP is still valid.

+

Updated the code to check all IP addresses and flag a change only if the + original IP is no longer valid.

+

Resolves: #924

+

geolocation.sample.conf: Fix comment marker at end of file

+

Author: George Joseph + Date: 2024-10-08

+

Resolves: #937

+

func_base64.c: Ensure we set aside enough room for base64 encoded data.

+

Author: Sean Bright + Date: 2024-10-08

+

Reported by SingularTricycle on IRC.

+

Fixes #940

+

app_dial: Fix progress timeout.

+

Author: Naveen Albert + Date: 2024-10-03

+

Under some circumstances, the progress timeout feature added in commit + 320c98eec87c473bfa814f76188a37603ea65ddd does not work as expected, + such as if there is no media flowing. Adjust the waitfor call to + explicitly use the progress timeout if it would be reached sooner than + the answer timeout to ensure we handle the timers properly.

+

Resolves: #821

+

chan_dahdi: Never send MWI while off-hook.

+

Author: Naveen Albert + Date: 2024-10-01

+

In some circumstances, it is possible for the do_monitor thread to + erroneously think that a line is on-hook and send an MWI FSK spill + to it when the line is really off-hook and no MWI should be sent. + Commit 0a8b3d34673277b70be6b0e8ac50191b1f3c72c6 previously fixed this + issue in a more readily encountered scenario, but it has still been + possible for MWI to be sent when it shouldn't be. To robustly fix + this issue, query DAHDI for the hook status to ensure we don't send + MWI on a line that is actually still off hook.

+

Resolves: #928

+

manager.c: Add unit test for Originate app and appdata permissions

+

Author: George Joseph + Date: 2024-10-03

+

This unit test checks that dialplan apps and app data specified + as parameters for the Originate action are allowed with the + permissions the user has.

+

alembic: Drop redundant voicemail_messages index.

+

Author: Sean Bright + Date: 2024-09-26

+

The voicemail_messages_dir index is a left prefix of the table's + primary key and therefore unnecessary.

+

res_agi.c: Ensure SIGCHLD handler functions are properly balanced.

+

Author: Sean Bright + Date: 2024-09-30

+

Calls to ast_replace_sigchld() and ast_unreplace_sigchld() must be + balanced to ensure that we can capture the exit status of child + processes when we need to. This extends to functions that call + ast_replace_sigchld() and ast_unreplace_sigchld() such as + ast_safe_fork() and ast_safe_fork_cleanup().

+

The primary change here is ensuring that we do not call + ast_safe_fork_cleanup() in res_agi.c if we have not previously + called ast_safe_fork().

+

Additionally we reinforce some of the documentation and add an + assertion to, ideally, catch this sooner were this to happen again.

+

Fixes #922

+

main, res, tests: Fix compilation errors on FreeBSD.

+

Author: Naveen Albert + Date: 2024-09-29

+

asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings. + config.c: Include header file for WIFEXITED/WEXITSTATUS macros. + res_timing_kqueue: Use more portable format specifier. + test_crypto: Use non-linux limits.h header file.

+

Resolves: #916

+

res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs

+

Author: George Joseph + Date: 2024-09-16

+

In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned + success but with a timeout of 0, we were stopping the timer and + decrementing the refcount on instance but not resetting the + timeout_timer to -1. When dtls_srtp_stop_timeout_timer() + was later called, it was atempting to stop a stale timer and could + decrement the refcount on instance again which would then cause + the instance destructor to run early. This would result in either + a FRACK or a SEGV when ast_rtp_stop(0 was called.

+

According to the OpenSSL docs, we shouldn't have been stopping the + timer when DTLSv1_get_timeout() returned success and the new timeout + was 0 anyway. We should have been calling DTLSv1_handle_timeout() + again immediately so we now reschedule the timer callback for + 1ms (almost immediately).

+

Additionally, instead of scheduling the timer callback at a fixed + interval returned by the initial call to DTLSv1_get_timeout() + (usually 999 ms), we now reschedule the next callback based on + the last call to DTLSv1_get_timeout().

+

Resolves: #487

+

manager.c: Restrict ModuleLoad to the configured modules directory.

+

Author: Ben Ford + Date: 2024-09-25

+

When using the ModuleLoad AMI action, it was possible to traverse + upwards through the directories to files outside of the configured + modules directory. We decided it would be best to restrict access to + modules exclusively in the configured directory. You will now get an + error when the specified module is outside of this limitation.

+

Fixes: #897

+

UserNote: The ModuleLoad AMI action now restricts modules to the + configured modules directory.

+

res_agi.c: Prevent possible double free during SPEECH RECOGNIZE

+

Author: jiangxc + Date: 2024-07-17

+

When using the speech recognition module, crashes can occur + sporadically due to a "double free or corruption (out)" error. Now, in + the section where the audio stream is being captured in a loop, each + time after releasing fr, it is set to NULL to prevent repeated + deallocation.

+

Fixes #772

+

cdr_custom: Allow absolute filenames.

+

Author: Sean Bright + Date: 2024-09-26

+

A follow up to #893 that brings the same functionality to + cdr_custom. Also update the sample configuration files to note support + for absolute paths.

+

astfd.c: Avoid calling fclose with NULL argument.

+

Author: Naveen Albert + Date: 2024-09-24

+

Don't pass through a NULL argument to fclose, which is undefined + behavior, and instead return -1 and set errno appropriately. This + also avoids a compiler warning with glibc 2.38 and newer, as glibc + commit 71d9e0fe766a3c22a730995b9d024960970670af + added the nonnull attribute to this argument.

+

Resolves: #900

+

channel: Preserve CHANNEL(userfield) on masquerade.

+

Author: Peter Jannesen + Date: 2024-09-20

+

In certain circumstances a channel may undergo an operation + referred to as a masquerade. If this occurs the CHANNEL(userfield) + value was not preserved causing it to get lost. This change makes + it so that this field is now preserved.

+

Fixes: #882

+

cel_custom: Allow absolute filenames.

+

Author: Peter Jannesen + Date: 2024-09-20

+

If a filename starts with a '/' in cel_custom [mappings] assume it is + a absolute file path and not relative filename/path to + AST_LOG_DIR/cel_custom/

+

app_voicemail: Fix ill-formatted pager emails with custom subject.

+

Author: Naveen Albert + Date: 2024-09-24

+

Add missing end-of-headers newline to pager emails with custom + subjects, since this was missing from this code path.

+

Resolves: #902

+

res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery

+

Author: Sean Bright + Date: 2024-09-23

+

Fixes #895

+

Fix application references to Background

+

Author: George Joseph + Date: 2024-09-20

+

The app is actually named "BackGround" but several references + in XML documentation were spelled "Background" with the lower + case "g". This was causing documentation links to return + "not found" messages.

+

manager.conf.sample: Fix mathcing typo

+

Author: George Joseph + Date: 2024-09-24

+

manager: Enhance event filtering for performance

+

Author: George Joseph + Date: 2024-07-31

+

UserNote: You can now perform more granular filtering on events + in manager.conf using expressions like + eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/ + This is much more efficient than + eventfilter = Event: Newchannel.*Channel: PJSIP/ + Full syntax guide is in configs/samples/manager.conf.sample.

+

manager.c: Split XML documentation to manager_doc.xml

+

Author: George Joseph + Date: 2024-08-01

+

db.c: Remove limit on family/key length

+

Author: George Joseph + Date: 2024-09-11

+

Consumers like media_cache have been running into issues with + the previous astdb "/family/key" limit of 253 bytes when needing + to store things like long URIs. An Amazon S3 URI is a good example + of this. Now, instead of using a static 256 byte buffer for + "/family/key", we use ast_asprintf() to dynamically create it.

+

Both test_db.c and test_media_cache.c were also updated to use + keys/URIs over the old 253 character limit.

+

Resolves: #881

+

UserNote: The ast_db_*() APIs have had the 253 byte limit on + "/family/key" removed and will now accept families and keys with a + total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This + affects the DB* dialplan applications, dialplan functions, + manager actions and databse CLI commands. Since the + media_cache also uses the ast_db_*() APIs, you can now store + resources with URIs longer than 253 bytes.

+

stir_shaken: Fix propagation of attest_level and a few other values

+

Author: George Joseph + Date: 2024-09-24

+

attest_level, send_mky and check_tn_cert_public_url weren't + propagating correctly from the attestation object to the profile + and tn.

+ +

Resolves: #904

+

res_stir_shaken: Remove stale include for jansson.h in verification.c

+

Author: George Joseph + Date: 2024-09-17

+

verification.c had an include for jansson.h left over from previous + versions of the module. Since res_stir_shaken no longer has a + dependency on jansson, the bundled version wasn't added to GCC's + include path so if you didn't also have a jansson development package + installed, the compile would fail. Removing the stale include + was the only thing needed.

+

Resolves: #889

+

res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid

+

Author: George Joseph + Date: 2024-09-13

+ +

Resolves: #884

+

res_stir_shaken: Check for disabled before param validation

+

Author: George Joseph + Date: 2024-09-11

+

For both attestation and verification, we now check whether they've + been disabled either globally or by the profile before validating + things like callerid, orig_tn, dest_tn, etc. This prevents useless + error messages.

+

Resolves: #879

+

app_chanspy.c: resolving the issue writing frame to whisper audiohook.

+

Author: Tinet-mucw + Date: 2024-09-10

+

ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.

+

Resolves: #876

+

autoservice: Do not sleep if autoservice_stop is called within autoservice thread

+

Author: Alexei Gradinari + Date: 2024-09-04

+

It's possible that ast_autoservice_stop is called within the autoservice thread. + In this case the autoservice thread is stuck in an endless sleep.

+

To avoid endless sleep ast_autoservice_stop must check that it's not called + within the autoservice thread.

+

Fixes: #763

+

res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback

+

Author: George Joseph + Date: 2024-08-12

+

The ub_result pointer passed to unbound_resolver_callback by + libunbound can be NULL if the query was for something malformed + like .1 or [.1]. If it is, we now set a 'ns_r_formerr' result + and return instead of crashing with a SEGV. This causes pjproject + to simply cancel the transaction with a "No answer record in the DNS + response" error. The existing "off nominal" unit test was also + updated to check this condition.

+

Although not necessary for this fix, we also made + ast_dns_resolver_completed() tolerant of a NULL result.

+

Resolves: GHSA-v428-g3cw-7hv9

+

app_voicemail: Use ast_asprintf to create mailbox SQL query

+

Author: George Joseph + Date: 2024-09-03

+

...instead of trying to calculate the length of the buffer needed + manually.

+

res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch

+

Author: Mike Bradeen + Date: 2024-08-21

+

When Asterisk sends an offer to Bob that includes 48K and 8K codecs with + matching 4733 offers, Bob may want to use the 48K audio codec but can not + accept 48K digits and so negotiates for a mixed set.

+

Asterisk will now check Bob's offer to make sure Bob has indicated this is + acceptible and if not, will use Bob's preference.

+

Fixes: #847

+

app_chanspy.c: resolving the issue with audiohook direction read

+

Author: Tinet-mucw + Date: 2024-08-30

+

ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.

+

Resolves: #861

+

security_agreements.c: Refactor the to_str functions and fix a few other bugs

+

Author: George Joseph + Date: 2024-08-17

+ +

Resolves: #845

+

res_pjsip_sdp_rtp fix leaking astobj2 ast_format

+

Author: Alexei Gradinari + Date: 2024-08-23

+

PR #700 added a preferred_format for the struct ast_rtp_codecs, + but when set the preferred_format it leaks an astobj2 ast_format. + In the next code + ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0)); + both functions ast_rtp_codecs_set_preferred_format + and ast_format_cap_get_format increases the ao2 reference count.

+

Fixes: #856

+

stir_shaken.conf.sample: Fix bad references to private_key_path

+

Author: George Joseph + Date: 2024-08-22

+

They should be private_key_file.

+

Resolves: #854

+

res_pjsip_logger.c: Fix 'OPTIONS' tab completion.

+

Author: Sean Bright + Date: 2024-08-19

+

Fixes #843

+

alembic: Make 'revises' header comment match reality.

+

Author: Sean Bright + Date: 2024-08-17

+

Update version for Asterisk 22

+

Author: Mike Bradeen + Date: 2024-08-14

+

chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay

+

Author: Cade Parker + Date: 2024-08-07

+

On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)

+

res_pjsip_notify: add dialplan application

+

Author: Mike Bradeen + Date: 2024-07-09

+

Add dialplan application PJSIPNOTIFY to send either pre-configured + NOTIFY messages from pjsip_notify.conf or with headers defined in + dialplan.

+

Also adds the ability to send pre-configured NOTIFY commands to a + channel via the CLI.

+

Resolves: #799

+

UserNote: A new dialplan application PJSIPNotify is now available + which can send SIP NOTIFY requests from the dialplan.

+

The pjsip send notify CLI command has also been enhanced to allow + sending NOTIFY messages to a specific channel. Syntax:

+

pjsip send notify

+

manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE

+

Author: George Joseph + Date: 2024-08-08

+

If you run an AMI CoreShowChannelMap on a channel that isn't in a + bridge and you're in DEVMODE, you can get a FRACK because the + bridge id is empty. We now simply return an empty list for that + request.

+

channel: Add multi-tenant identifier.

+

Author: Ben Ford + Date: 2024-05-21

+

This patch introduces a new identifier for channels: tenantid. It's + a stringfield on the channel that can be used for general purposes. It + will be inherited by other channels the same way that linkedid is.

+

You can set tenantid in a few ways. The first is to set it in the + dialplan with the Set and CHANNEL functions:

+

exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)

+

It can also be accessed via CHANNEL:

+

exten => example,2,NoOp(CHANNEL(tenantid))

+

Another method is to use the new tenantid option for pjsip endpoints in + pjsip.conf:

+

[my_endpoint] + type=endpoint + tenantid=My tenant ID

+

This is considered the best approach since you will be able to see the + tenant ID as early as the Newchannel event.

+

It can also be set using set_var in pjsip.conf on the endpoint like + setting other channel variable:

+

set_var=CHANNEL(tenantid)=My tenant ID

+

Note that set_var will not show tenant ID on the Newchannel event, + however.

+

Tenant ID has also been added to CDR. It's read-only and can be accessed + via CDR(tenantid). You can also get the tenant ID of the last channel + communicated with via CDR(peertenantid).

+

Tenant ID will also show up in CEL records if it has been set, and the + version number has been bumped accordingly.

+

Fixes: #740

+

UserNote: tenantid has been added to channels. It can be read in + dialplan via CHANNEL(tenantid), and it can be set using + Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to + use the new tenantid option for pjsip endpoints (e.g., tenantid=My + tenant ID) so that it will show up in Newchannel events. You can set it + like any other channel variable using set_var in pjsip.conf as well, but + note that this will NOT show up in Newchannel events. Tenant ID is also + available in CDR and can be accessed with CDR(tenantid). The peer tenant + ID can also be accessed with CDR(peertenantid). CEL includes tenant ID + as well if it has been set.

+

UpgradeNote: A new versioned struct (ast_channel_initializers) has been + added that gets passed to __ast_channel_alloc_ap. The new function + ast_channel_alloc_with_initializers should be used when creating + channels that require the use of this struct. Currently the only value + in the struct is for tenantid, but now more fields can be added to the + struct as necessary rather than the __ast_channel_alloc_ap function. A + new option (tenantid) has been added to endpoints in pjsip.conf as well. + CEL has had its version bumped to include tenant ID.

+

configure: Use . file rather than source file.

+

Author: Jaco Kroon + Date: 2024-08-05

+

source is a bash concept, so when /bin/sh points to another shell the + existing construct won't work.

+

Reference: https://bugs.gentoo.org/927055 + Signed-off-by: Jaco Kroon jaco@uls.co.za

+

feat: ARI "ChannelToneDetected" event

+

Author: gibbz00 + Date: 2024-07-18

+

A stasis event is now produced when using the TONE_DETECT dialplan + function. This event is published over ARI using the ChannelToneDetected + event. This change does not make it available over AMI.

+

Fixes: #811

+

UserNote: Setting the TONE_DETECT dialplan function on a channel + in ARI will now cause a ChannelToneDetected ARI event to be raised + when the specified tone is detected.

+

manager.c: Add entries to Originate blacklist

+

Author: George Joseph + Date: 2024-07-22

+

Added Reload and DBdeltree to the list of dialplan application that + can't be executed via the Originate manager action without also + having write SYSTEM permissions.

+

Added CURL, DB, FILE, ODBC and REALTIME to the list of dialplan + functions that can't be executed via the Originate manager action + without also having write SYSTEM permissions.

+

If the Queue application is attempted to be run by the Originate + manager action and an AGI parameter is specified in the app data, + it'll be rejected unless the manager user has either the AGI or + SYSTEM permissions.

+

Resolves: #GHSA-c4cg-9275-6w44

+

res_stasis: fix intermittent delays on adding channel to bridge

+

Author: Mike Bradeen + Date: 2024-07-10

+

Previously, on command execution, the control thread was awoken by + sending a SIGURG. It was found that this still resulted in some + instances where the thread was not immediately awoken.

+

This change instead sends a null frame to awaken the control thread, + which awakens the thread more consistently.

+

Resolves: #801

+

res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto

+

Author: Tinet-mucw + Date: 2024-08-02

+

When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.

+

Resolves: asterisk#826

+

rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()

+

Author: George Joseph + Date: 2024-07-25

+

There can be empty slots in payload_mapping_tx corresponding to + dynamic payload types that haven't been seen before so we now + check for NULL before attempting to use 'type' in the call to + ast_format_cmp.

+

Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()

+

Resolves: #822

+

stir_shaken: CRL fixes and a new CLI command

+

Author: George Joseph + Date: 2024-07-19

+ +

Resolves: #809

+

res_pjsip_config_wizard.c: Refactor load process

+

Author: George Joseph + Date: 2024-07-23

+

The way we have been initializing the config wizard prevented it + from registering its objects if res_pjsip happened to load + before it.

+ +

Resolves: #816

+

UserNote: The res_pjsip_config_wizard.so module can now be reloaded.

+

voicemail.conf.sample: Fix ':' comment typo

+

Author: George Joseph + Date: 2024-07-24

+

...and removed an errant trailing space.

+

Resolves: #819

+

bridge_softmix: Fix queueing VIDUPDATE control frames

+

Author: George Joseph + Date: 2024-07-17

+

softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else() + with the bridge_channel so the VIDUPDATE control frame isn't echoed back.

+

softmix_bridge_write_control() was setting bridge_channel to NULL + when calling ast_bridge_queue_everyone_else() for VIDUPDATE control + frames. This was causing the frame to be echoed back to the + channel it came from. In certain cases, like when two channels or + bridges are being recorded, this can cause a ping-pong effect that + floods the system with VIDUPDATE control frames.

+

Resolves: #780

+

res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()

+

Author: Igor Goncharovsky + Date: 2024-05-12

+

When using the PJSIP_DIAL_CONTACTS() function for use in the Dial() + command, the contacts are returned in text form, so the input to + the path_outgoing_request() function is a contact value of NULL. + The issue was reported in ASTERISK-28211, but was not actually fixed + in ASTERISK-30100. This fix brings back the code that was previously + removed and adds code to search for a contact to extract the path + value from it.

+

res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits

+

Author: Mike Bradeen + Date: 2024-06-21

+

After change made in 624f509 to add support for non 8K RFC 4733/2833 digits, + Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of + the negotiated codec(s).

+

This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC + offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.

+

A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to + be re-written to allow for these scenarios.

+

Fixes: #776

+

ast-db-manage: Remove duplicate enum creation

+

Author: George Joseph + Date: 2024-07-08

+

Remove duplicate creation of ast_bool_values from + 2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was + causing alembic upgrades to fail since the enum was already created + in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.

+

Resolves: #797

+

security_agreement.c: Always add the Require and Proxy-Require headers

+

Author: George Joseph + Date: 2024-07-03

+

The Require: mediasec and Proxy-Require: mediasec headers need + to be sent whenever we send Security-Client or Security-Verify + headers but the logic to do that was only in add_security_headers() + in res_pjsip_outbound_register. So while we were sending them on + REGISTER requests, we weren't sending them on INVITE requests.

+

This commit moves the logic to send the two headers out of + res_pjsip_outbound_register:add_security_headers() and into + security_agreement:ast_sip_add_security_headers(). This way + they're always sent when we send Security-Client or + Security-Verify.

+

Resolves: #789

+

logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.

+

Author: Sean Bright + Date: 2024-06-29

+

Fixes #785

+

stasis_channels: Use uniqueid and name to delete old snapshots

+

Author: George Joseph + Date: 2024-05-08

+

Whenver a new channel snapshot is created or when a channel is + destroyed, we need to delete any existing channel snapshot from + the snapshot cache. Historically, we used the channel->snapshot + pointer to delete any existing snapshots but this has two issues.

+

First, if something (possibly ast_channel_internal_swap_snapshots) + sets channel->snapshot to NULL while there's still a snapshot in + the cache, we wouldn't be able to delete it and it would be orphaned + when the channel is destroyed. Since we use the cache to list + channels from the CLI, AMI and ARI, it would appear as though the + channel was still there when it wasn't.

+

Second, since there are actually two caches, one indexed by the + channel's uniqueid, and another indexed by the channel's name, + deleting from the caches by pointer requires a sequential search of + all of the hash table buckets in BOTH caches to find the matching + snapshots. Not very efficient.

+

So, we now delete from the caches using the channel's uniqueid + and name. This solves both issues.

+

This doesn't address how channel->snapshot might have been set + to NULL in the first place because although we have concrete + evidence that it's happening, we haven't been able to reproduce it.

+

Resolves: #783

+

app_voicemail_odbc: Allow audio to be kept on disk

+

Author: George Joseph + Date: 2024-04-09

+

This commit adds a new voicemail.conf option 'odbc_audio_on_disk' + which when set causes the ODBC variant of app_voicemail to leave + the message and greeting audio files on disk and only store the + message metadata in the database. This option came from a concern + that the database could grow to large and cause remote access + and/or replication to become slow. In a clustering situation + with this option, all asterisk instances would share the same + database for the metadata and either use a shared filesystem + or other filesystem replication service much more suitable + for synchronizing files.

+

The changes to app_voicemail to implement this feature were actually + quite small but due to the complexity of the module, the actual + source code changes were greater. They fall into the following + categories:

+ +

UserNote: This commit adds a new voicemail.conf option + 'odbc_audio_on_disk' which when set causes the ODBC variant of + app_voicemail_odbc to leave the message and greeting audio files + on disk and only store the message metadata in the database. + Much more information can be found in the voicemail.conf.sample + file.

+

bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while iterating over bridge->channels. From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition.

+

Author: Tinet-mucw + Date: 2024-06-13

+

Resolves: https://github.com/asterisk/asterisk/issues/768

+

app_queue: Add option to not log Restricted Caller ID to queue_log

+

Author: Alexei Gradinari + Date: 2024-06-12

+

Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event + in the queue log if the Caller ID is restricted.

+

Resolves: #765

+

UpgradeNote: Add a new column to the queues table: + queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') + to control whether the Restricted Caller ID will be stored in the queue log.

+

UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID + will be stored in the queue log. + If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.

+

pbx.c: expand fields width of "core show hints"

+

Author: Alexei Gradinari + Date: 2024-06-13

+

The current width for "extension" is 20 and "device state id" is 20, which is too small. + The "extension" field contains "ext"@"context", so 20 characters is not enough. + The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.

+

Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.

+

Resolves: #770

+

UserNote: The fields width of "core show hints" were increased. + The width of "extension" field to 30 characters and + the width of the "device state id" field to 60 characters.

+

pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.

+

Author: Sean Bright + Date: 2024-06-02

+

Various SIP headers permit a URI to be prefaced with a display-name + production that can include characters (like commas and parentheses) + that are problematic for Asterisk's dialplan parser and, specifically + in the case of this patch, the PJSIP_PARSE_URI function.

+

This patch introduces a new function - PJSIP_PARSE_URI_FROM - that + behaves identically to PJSIP_PARSE_URI except that the first + argument is now a variable name and not a literal URI.

+

Fixes #756

+

manager.c: Properly terminate CoreShowChannelMap event.

+

Author: Sean Bright + Date: 2024-06-10

+

Fixes #761

+

cli: Show configured cache dir

+

Author: Bastian Triller + Date: 2024-06-07

+

Since Asterisk 19 it is possible to cache recorded files into another + directory [1] [2]. + Show configured location of cache dir in CLI's core show settings.

+

[1] ASTERISK-29143 + [2] b08427134fd51bb549f198e9f60685f2680c68d7

+

xml.c: Update deprecated libxml2 API usage.

+

Author: Sean Bright + Date: 2024-05-23

+

Two functions are deprecated as of libxml2 2.12:

+
* xmlSubstituteEntitiesDefault
+* xmlParseMemory
+
+

So we update those with supported API.

+

Additionally, res_calendar_caldav has been updated to use libxml2's + xmlreader API instead of the SAX2 API which has always felt a little + hacky (see deleted comment block in res_calendar_caldav.c).

+

The xmlreader API has been around since libxml2 2.5.0 which was + released in 2003.

+

Fixes #725

+

cdr_pgsql: Fix crash when the module fails to load multiple times.

+

Author: chrsmj + Date: 2024-05-16

+

Missing or corrupt cdr_pgsql.conf configuration file can cause the + second attempt to load the PostgreSQL CDR module to crash Asterisk via + the Command Line Interface because a null CLI command is registered on + the first failed attempt to load the module.

+

Resolves: #736

+

asterisk.c: Don't log an error if .asterisk_history does not exist.

+

Author: Sean Bright + Date: 2024-05-27

+

Fixes #751

+

chan_ooh323: Fix R/0 typo in docs

+

Author: Walter Doekes + Date: 2024-05-27

+

bundled_pjproject: Disable UPnP support.

+

Author: Sean Bright + Date: 2024-05-24

+

Fixes #747

+

file.h: Rename function argument to avoid C++ keyword clash.

+

Author: Sean Bright + Date: 2024-05-24

+

Fixes #744

+

rtp_engine: add support for multirate RFC2833 digits

+

Author: Mike Bradeen + Date: 2024-04-08

+

Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.

+

Asterisk currently treats RFC2833 Digits as a single rtp payload type + with a fixed bitrate of 8K. This change would expand that to 8, 16, + 24 and 32K.

+

This requires checking the offered rtp types for any of these bitrates + and then adding an offer for each (if configured for RFC2833.) DTMF + generation must also be changed in order to look at the current outbound + codec in order to generate appropriately timed rtp.

+

For cases where no outgoing audio has yet been sent prior to digit + generation, Asterisk now has a concept of a 'preferred' codec based on + offer order.

+

On inbound calls Asterisk will mimic the payload types of the RFC2833 + digits.

+

On outbound calls Asterisk will choose the next free payload types starting + with 101.

+

UserNote: No change in configuration is required in order to enable this + feature. Endpoints configured to use RFC2833 will automatically have this + enabled. If the endpoint does not support this, it should not include it in + the SDP offer/response.

+

Resolves: #699

+

configs: Fix a misleading IPv6 ACL example in Named ACLs

+

Author: Ivan Poddubny + Date: 2024-05-05

+

"deny=::" is equivalent to "::/128". + In order to mean "deny everything by default" it must be "::/0".

+

asterisk.c: Fix sending incorrect messages to systemd notify

+

Author: Ivan Poddubny + Date: 2024-05-05

+

Send "RELOADING=1" instead of "RELOAD=1" to follow the format + expected by systemd (see sd_notify(3) man page).

+

Do not send STOPPING=1 in remote console mode: + attempting to execute "asterisk -rx" by the main process leads to + a warning if NotifyAccess=main (the default) or to a forced termination + if NotifyAccess=all.

+

res/stasis/control.c: include signal.h

+

Author: Fabrice Fontaine + Date: 2024-05-01

+

Include signal.h to avoid the following build failure with uclibc-ng + raised since + https://github.com/asterisk/asterisk/commit/2694792e13c7f3ab1911c4a69fba0df32c544177:

+

stasis/control.c: In function 'exec_command_on_condition': + stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration] + 313 | pthread_kill(control->control_thread, SIGURG); + | ^~~~~~~~~~~~ + | pthread_yield + stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function) + 313 | pthread_kill(control->control_thread, SIGURG); + | ^~~~~~

+

cherry-pick-to: 18 + cherry-pick-to: 20 + cherry-pick-to: 21

+

Fixes: #729

+

res_pjsip_logger: Preserve logging state on reloads.

+

Author: Naveen Albert + Date: 2023-08-09

+

Currently, reloading res_pjsip will cause logging + to be disabled. This is because logging can also + be controlled via the debug option in pjsip.conf + and this defaults to "no".

+

To improve this, logging is no longer disabled on + reloads if logging had not been previously + enabled using the debug option from the config. + This ensures that logging enabled from the CLI + will persist through a reload.

+

ASTERISK-29912 #close

+

Resolves: #246

+

UserNote: Issuing "pjsip reload" will no longer disable + logging if it was previously enabled from the CLI.

+

logger: Add unique verbose prefixes for levels 5-10.

+

Author: Naveen Albert + Date: 2024-04-27

+

Add unique verbose prefixes for levels higher than 4, so + that these can be visually differentiated from each other.

+

Resolves: #721

+

say.c: Fix cents off-by-one due to floating point rounding.

+

Author: Naveen Albert + Date: 2024-01-10

+

Some of the money announcements can be off by one cent, + due to the use of floating point in the money calculations, + which is bad for obvious reasons.

+

This replaces floating point with simple string parsing + to ensure the cents value is converted accurately.

+

Resolves: #525

+

loader.c: Allow dependent modules to be unloaded recursively.

+

Author: Naveen Albert + Date: 2023-12-02

+

Because of the (often recursive) nature of module dependencies in + Asterisk, hot swapping a module on the fly is cumbersome if a module + is depended on by other modules. Currently, dependencies must be + popped manually by unloading dependents, unloading the module of + interest, and then loading modules again in reverse order.

+

To make this easier, the ability to do this recursively in certain + circumstances has been added, as an optional extension to the + "module refresh" command. If requested, Asterisk will check if a module + that has a positive usecount could be unloaded safely if anything + recursively dependent on it were unloaded. If so, it will go ahead + and unload all these modules and load them back again. This makes + hot swapping modules that provide dependencies much easier.

+

Resolves: #474

+

UserNote: In certain circumstances, modules with dependency relations + can have their dependents automatically recursively unloaded and loaded + again using the "module refresh" CLI command or the ModuleLoad AMI command.

+

res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting.

+

Author: Henrik Liljedahl + Date: 2024-04-11

+

First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.

+

Fixes #710

+

tcptls/iostream: Add support for setting SNI on client TLS connections

+

Author: George Joseph + Date: 2024-04-23

+

If the hostname field of the ast_tcptls_session_args structure is + set (which it is for websocket client connections), that hostname + will now automatically be used in an SNI TLS extension in the client + hello.

+

Resolves: #713

+

UserNote: Secure websocket client connections now send SNI in + the TLS client hello.

+

stir_shaken: Fix memory leak, typo in config, tn canonicalization

+

Author: George Joseph + Date: 2024-04-25

+ +

Resolves: #716

+

make_buildopts_h: Always include DETECT_DEADLOCKS

+

Author: George Joseph + Date: 2024-04-27

+

Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must + always be included in buildopts.h instead of only when + ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise.

+

Resolves: #719

+

sorcery.c: Fixed crash error when executing "module reload"

+

Author: Spiridonov Dmitry + Date: 2024-04-14

+

Fixed crash error when cli "module reload". The error appears when + compiling with res_prometheus and using the sorcery memory cache for + registrations

+

callerid.c: Parse previously ignored Caller ID parameters.

+

Author: Naveen Albert + Date: 2024-04-01

+

Commit f2f397c1a8cc48913434ebb297f0ff50d96993db previously + made it possible to send Caller ID parameters to FXS stations + which, prior to that, could not be sent.

+

This change is complementary in that we now handle receiving + all these parameters on FXO lines and provide these up to + the dialplan, via chan_dahdi. In particular:

+ +

With this change, Asterisk's Caller ID implementation is now + reasonably complete for both FXS and FXO operation.

+

Resolves: #681

+

logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT

+

Author: George Joseph + Date: 2024-04-09

+

If you're tracing a large function that may call another function + multiple times in different circumstances, it can be difficult to + see from the trace output exactly which location that function + was called from. There's no good way to automatically determine + the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT + simply print out a trace line before and after the call.

+

The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is + that SCOPE_CALL ignores the function's return value (if any) where + SCOPE_CALL_WITH_RESULT allows you to specify the type of the + function's return value so it can be assigned to a variable. + SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT + and the "int" return type.

+

app_queue.c: Properly handle invalid strategies from realtime.

+

Author: Sean Bright + Date: 2024-04-13

+

The existing code sets the queue strategy to ringall but it is then + immediately overwritten with an invalid one.

+

Fixes #707

+

file.c, channel.c: Don't emit warnings if progress received.

+

Author: Naveen Albert + Date: 2024-04-09

+

Silently ignore AST_CONTROL_PROGRESS where appropriate, + as most control frames already are.

+

Resolves: #696

+

alembic: Correct NULLability of PJSIP id columns.

+

Author: Sean Bright + Date: 2024-04-06

+

Fixes #695

+

rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup

+

Author: George Joseph + Date: 2024-04-02

+

rtp_engine.c and stun.c were calling ast_register_cleanup which + is skipped if any loadable module can't be cleanly unloaded + when asterisk shuts down. Since this will always be the case, + their cleanup functions never get run. In a practical sense + this makes no difference since asterisk is shutting down but if + you're in development mode and trying to use the leak sanitizer, + the leaks from both of those modules clutter up the output.

+

manager.c: Add missing parameters to Login documentation

+

Author: George Joseph + Date: 2024-04-03

+ +

Resolves: #689

+

func_callerid: Emit warning if invalid redirecting reason set.

+

Author: Naveen Albert + Date: 2024-04-01

+

Emit a warning if REDIRECTING(reason) is set to an invalid + reason, consistent with what happens when + REDIRECTING(orig-reason) is set to an invalid reason.

+

Resolves: #683

+

chan_dahdi: Add DAHDIShowStatus AMI action.

+

Author: Naveen Albert + Date: 2024-03-29

+ +

Resolves: #673

+

res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2)

+

Author: George Joseph + Date: 2024-04-01

+ +

Resolves: #676

+

Fix incorrect application and function documentation references

+

Author: George Joseph + Date: 2024-04-01

+

There were a few references in the embedded documentation XML + where the case didn't match or where the referenced app or function + simply didn't exist any more. These were causing 404 responses + in docs.asterisk.org.

+

cli.c: core show channels concise is not really deprecated.

+

Author: Sean Bright + Date: 2024-04-01

+

Fixes #675

+

res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI

+

Author: Sperl Viktor + Date: 2024-03-28

+

Add ability to match against PJSIP request URI.

+

UserNote: this new feature let users match endpoints based on the + indound SIP requests' URI. To do so, add 'request_uri' to the + endpoint's 'identify_by' option. The 'match_request_uri' option of + the identify can be an exact match for the entire request uri, or a + regular expression (between slashes). It's quite similar to the + header identifer.

+

Fixes: #599

+

Implement Configurable TCP Keepalive Settings in PJSIP Transports

+

Author: Joshua Elson + Date: 2024-03-18

+

This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary.

+

Fixes #657

+

chan_dahdi: Don't retry opening nonexistent channels on restart.

+

Author: Naveen Albert + Date: 2024-03-26

+

Commit 729cb1d390b136ccc696430aa5c68d60ea4028be added logic to retry + opening DAHDI channels on "dahdi restart" if they failed initially, + up to 1,000 times in a loop, to address cases where the channel was + still in use. However, this retry loop does not use the actual error, + which means chan_dahdi will also retry opening nonexistent channels + 1,000 times per channel, causing a flood of unnecessary warning logs + for an operation that will never succeed, with tens or hundreds of + thousands of open attempts being made.

+

The original patch would have been more targeted if it only retried + on the specific relevant error (likely EBUSY, although it's hard to + say since the original issue is no longer available).

+

To avoid the problem above while avoiding the possibility of breakage, + this skips the retry logic if the error is ENXIO (No such device or + address), since this will never succeed.

+

Resolves: #669

+

res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA

+

Author: Martin Tomec + Date: 2024-02-06

+

There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip.

+

Fixes: #579

+

UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.

+

res_ari.c: Add additional output to ARI requests when debug is enabled

+

Author: Martin Nystroem + Date: 2024-03-22

+

When ARI debug is enabled the logs will now output http method and the uri.

+

Fixes: #666

+

alembic: Fix compatibility with SQLAlchemy 2.0+.

+

Author: Sean Bright + Date: 2024-03-20

+

SQLAlchemy 2.0 changed the way that commits/rollbacks are handled + causing the final UPDATE to our alembic_version_<whatever> tables + to be rolled back instead of committed.

+

We now use one connection to determine which + alembic_version_<whatever> table to use and another to run the + actual migrations. This prevents the erroneous rollback.

+

This change is compatible with both SQLAlchemy 1.4 and 2.0.

+

manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action

+

Author: jonatascalebe + Date: 2024-03-14

+

manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action

+

The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too.

+

For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient.

+

UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication: + [addautoanswer] + exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0) + exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0) + exten => _s,n,Return()

+

menuselect: Minor cosmetic fixes.

+

Author: Naveen Albert + Date: 2024-03-21

+

Improve some of the formatting from + dd3f17c699e320d6d30c94298d8db49573ba28da + (#521).

+

pbx_variables.c: Prevent SEGV due to stack overflow.

+

Author: Naveen Albert + Date: 2023-12-04

+

It is possible for dialplan to result in an infinite + recursion of variable substitution, which eventually + leads to stack overflow. If we detect this, abort + substitution and log an error for the user to fix + the broken dialplan.

+

Resolves: #480

+

UpgradeNote: The maximum amount of dialplan recursion + using variable substitution (such as by using EVAL_EXTEN) + is capped at 15.

+

res_prometheus: Fix duplicate output of metric and help text

+

Author: Holger Hans Peter Freyther + Date: 2024-02-24

+

The prometheus exposition format requires each line to be unique[1]. + This is handled by struct prometheus_metric having a list of children + that is managed when registering a metric. In case the scrape callback + is used, it is the responsibility of the implementation to handle this + correctly.

+

Originally the bridge callback didn't handle NULL snapshots, the crash + fix lead to NULL metrics, and fixing that lead to duplicates.

+

The original code assumed that snapshots are not NULL and then relied on + "if (i > 0)" to establish the parent/children relationship between + metrics of the same class. This is not workerable as the first bridge + might be invisible/lacks a snapshot.

+

Fix this by keeping a separate array of the first metric by class. + Instead of relying on the index of the bridge, check whether the array + has an entry. Use that array for the output.

+

Add a test case that verifies that the help text is not duplicated.

+

Resolves: #642

+

[1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting

+

manager.c: Add CLI command to kick AMI sessions.

+

Author: Naveen Albert + Date: 2023-12-06

+

This adds a CLI command that can be used to manually + kick specific AMI sessions.

+

Resolves: #485

+

UserNote: The "manager kick session" CLI command now + allows kicking a specified AMI session.

+

chan_dahdi: Allow specifying waitfordialtone per call.

+

Author: Naveen Albert + Date: 2023-12-02

+

The existing "waitfordialtone" setting in chan_dahdi.conf + applies permanently to a specific channel, regardless of + how it is being used. This rather restrictively prevents + a system from simultaneously being able to pick free lines + for outgoing calls while also allowing barge-in to a trunk + by some other arrangement.

+

This allows specifying "waitfordialtone" using the CHANNEL + function for only the next call that will be placed, allowing + significantly more flexibility in the use of trunk interfaces.

+

Resolves: #472

+

UserNote: "waitfordialtone" may now be specified for DAHDI + trunk channels on a per-call basis using the CHANNEL function.

+

res_parking: Fail gracefully if parking lot is full.

+

Author: Naveen Albert + Date: 2024-03-03

+

Currently, if a parking lot is full, bridge setup returns -1, + causing dialplan execution to terminate without TryExec. + However, such failures should be handled more gracefully, + the same way they are on other paths, as indicated by the + module's author, here:

+

http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html

+

Now, callers will hear the parking failure announcement, and dialplan + will continue, which is consistent with existing failure modes.

+

Resolves: #624

+

res_config_mysql.c: Support hostnames up to 255 bytes.

+

Author: Sean Bright + Date: 2024-03-18

+

Fixes #654

+

res_pjsip: Fix alembic downgrade for boolean columns.

+

Author: Sean Bright + Date: 2024-03-18

+

When downgrading, ensure that we don't touch columns that didn't + actually change during upgrade.

+

Upgrade bundled pjproject to 2.14.1

+

Author: Stanislav Abramenkov + Date: 2024-03-12

+

Fixes: asterisk#648

+

UserNote: Bundled pjproject has been upgraded to 2.14.1. For more + information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1

+

alembic: Quote new MySQL keyword 'qualify.'

+

Author: Sean Bright + Date: 2024-03-15

+

Fixes #651

+

res_pjsip_session: Reset pending_media_state->read_callbacks

+

Author: Maximilian Fridrich + Date: 2024-02-15

+

In handle_negotiated_sdp the pending_media_state->read_callbacks must be + reset before they are added in the SDP handlers in + handle_negotiated_sdp_session_media. Otherwise, old callbacks for + removed streams and file descriptors could be added to the channel and + Asterisk would poll on non-existing file descriptors.

+

Resolves: #611

+

res_pjsip_stir_shaken.c: Add checks for missing parameters

+

Author: George Joseph + Date: 2024-03-11

+ +

Resolves: #645

+

app_dial: Add dial time for progress/ringing.

+

Author: Naveen Albert + Date: 2024-02-08

+

Add a timeout option to control the amount of time + to wait if no early media is received before giving + up. This allows aborting early if the destination + is not being responsive.

+

Resolves: #588

+

UserNote: The timeout argument to Dial now allows + specifying the maximum amount of time to dial if + early media is not received.

+

app_voicemail: Properly reinitialize config after unit tests.

+

Author: Naveen Albert + Date: 2024-02-29

+

Most app_voicemail unit tests were not properly cleaning up + after themselves after running. This led to test mailboxes + lingering around in the system. It also meant that if any + unit tests in app_voicemail that create mailboxes were executed + and the module was not unloaded/loaded again prior to running + the test_voicemail_vm_info unit test, Asterisk would segfault + due to an attempt to copy a NULL string.

+

The load_config test did actually have logic to reinitialize + the config after the test. However, this did not work in practice + since load_config() would not reload the config since voicemail.conf + had not changed during the test; thus, additional logic has been + added to ensure that voicemail.conf is truly reloaded, after any + unit tests which modify the users list.

+

This prevents the SEGV due to invalid mailboxes lingering around, + and also ensures that the system state is restored to what it was + prior to the tests running.

+

Resolves: #629

+

app_queue.c : fix "queue add member" usage string

+

Author: Shaaah + Date: 2024-01-23

+

Fixing bracket placement in the "queue add member" cli usage string.

+

app_voicemail: Allow preventing mark messages as urgent.

+

Author: Naveen Albert + Date: 2024-02-24

+

This adds an option to allow preventing callers from leaving + messages marked as 'urgent'.

+

Resolves: #619

+

UserNote: The leaveurgent mailbox option can now be used to + control whether callers may leave messages marked as 'Urgent'.

+

res_pjsip: Use consistent type for boolean columns.

+

Author: Sean Bright + Date: 2024-02-27

+

This migrates the relevant schema objects from the ('yes', 'no') + definition to the ('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no') + one.

+

Fixes #617

+

attestation_config.c: Use ast_free instead of ast_std_free

+

Author: George Joseph + Date: 2024-03-05

+

In as_check_common_config, we were calling ast_std_free on + raw_key but raw_key was allocated with ast_malloc so it + should be freed with ast_free.

+

Resolves: #636

+

Makefile: Add stir_shaken/cache to directories created on install

+

Author: George Joseph + Date: 2024-03-04

+

The default location for the stir_shaken cache is + /var/lib/asterisk/keys/stir_shaken/cache but we were only creating + /var/lib/asterisk/keys/stir_shaken on istall. We now create + the cache sub-directory.

+

Resolves: #634

+

Stir/Shaken Refactor

+

Author: George Joseph + Date: 2023-10-26

+

Why do we need a refactor?

+

The original stir/shaken implementation was started over 3 years ago + when little was understood about practical implementation. The + result was an implementation that wouldn't actually interoperate + with any other stir-shaken implementations.

+

There were also a number of stir-shaken features and RFC + requirements that were never implemented such as TNAuthList + certificate validation, sending Reason headers in SIP responses + when verification failed but we wished to continue the call, and + the ability to send Media Key(mky) grants in the Identity header + when the call involved DTLS.

+

Finally, there were some performance concerns around outgoing + calls and selection of the correct certificate and private key. + The configuration was keyed by an arbitrary name which meant that + for every outgoing call, we had to scan the entire list of + configured TNs to find the correct cert to use. With only a few + TNs configured, this wasn't an issue but if you have a thousand, + it could be.

+

What's changed?

+ +

Resolves: #351 + Resolves: #46

+

UserNote: Asterisk's stir-shaken feature has been refactored to + correct interoperability, RFC compliance, and performance issues. + See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more + information.

+

UpgradeNote: The stir-shaken refactor is a breaking change but since + it's not working now we don't think it matters. The + stir_shaken.conf file has changed significantly which means that + existing ones WILL need to be changed. The stir_shaken.conf.sample + file in configs/samples/ has quite a bit more information. This is + also an ABI breaking change since some of the existing objects + needed to be changed or removed, and new ones added. Additionally, + if res_stir_shaken is enabled in menuselect, you'll need to either + have the development package for libjwt v1.15.3 installed or use + the --with-libjwt-bundled option with ./configure.

+

translate.c: implement new direct comp table mode

+

Author: Sebastian Jennen + Date: 2024-02-25

+

The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio. + This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).

+ +

Resolves: #601

+

README.md: Removed outdated link

+

Author: Shyju Kanaprath + Date: 2024-02-23

+

Removed outdated link http://www.quicknet.net from README.md

+

cherry-pick-to: 18 + cherry-pick-to: 20 + cherry-pick-to: 21

+

strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.

+

Author: Sean Bright + Date: 2024-02-17

+

If a dynamic string is created with an initial length of 0, + ast_str_buffer(…) will return an invalid pointer.

+

This was a secondary discovery when fixing #65.

+

res_rtp_asterisk.c: Correct coefficient in MOS calculation.

+

Author: romryz + Date: 2024-02-06

+

Media Experience Score relies on incorrect pseudo_mos variable + calculation. According to forming an opinion section of the + documentation, calculation relies on ITU-T G.107 standard:

+
  https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion
+
+

ITU-T G.107 Annex B suggests to calculate MOS with a coefficient + "seven times ten to the power of negative six", 7 * 10^(-6). which + would mean 6 digits after the decimal point. Current implementation + has 7 digits after the decimal point, which downrates the calls.

+

Fixes: #597

+

dsp.c: Fix and improve potentially inaccurate log message.

+

Author: Naveen Albert + Date: 2024-02-09

+

If ast_dsp_process is called with a codec besides slin, ulaw, + or alaw, a warning is logged that in-band DTMF is not supported, + but this message is not always appropriate or correct, because + ast_dsp_process is much more generic than just DTMF detection.

+

This logs a more generic message in those cases, and also improves + codec-mismatch logging throughout dsp.c by ensuring incompatible + codecs are printed out.

+

Resolves: #595

+

pjsip show channelstats: Prevent possible segfault when faxing

+

Author: George Joseph + Date: 2024-02-09

+

Under rare circumstances, it's possible for the original audio + session in the active_media_state default_session to be corrupted + instead of removed when switching to the t38/image media session + during fax negotiation. This can cause a segfault when a "pjsip + show channelstats" attempts to print that audio media session's + rtp statistics. In these cases, the active_media_state + topology is correctly showing only a single t38/image stream + so we now check that there's an audio stream in the topology + before attempting to use the audio media session to get the rtp + statistics.

+

Resolves: #592

+

Reduce startup/shutdown verbose logging

+

Author: George Joseph + Date: 2024-01-31

+

When started with a verbose level of 3, asterisk can emit over 1500 + verbose message that serve no real purpose other than to fill up + logs. When asterisk shuts down, it emits another 1100 that are of + even less use. Since the testsuite runs asterisk with a verbose + level of 3, and asterisk starts and stops for every one of the 700+ + tests, the number of log messages is staggering. Besides taking up + resources, it also makes it hard to debug failing tests.

+

This commit changes the log level for those verbose messages to 5 + instead of 3 which reduces the number of log messages to only a + handful. Of course, NOTICE, WARNING and ERROR message are + unaffected.

+

There's also one other minor change... + ast_context_remove_extension_callerid2() logs a DEBUG message + instead of an ERROR if the extension you're deleting doesn't exist. + The pjsip_config_wizard calls that function to clean up the config + and has been triggering that annoying error message for years.

+

Resolves: #582

+

configure: Rerun bootstrap on modern platform.

+

Author: Naveen Albert + Date: 2024-02-08

+

The last time configure was run, it was run on a system that + did not enable -std=gnu11 by default, which meant that the + restrict qualifier would not be recognized on certain platforms. + This regenerates the configure files from running bootstrap.sh, + so that these should be recognized on all supported platforms.

+

Resolves: #586

+

Upgrade bundled pjproject to 2.14.

+

Author: Ben Ford + Date: 2024-02-05

+

Fixes: #406

+

UserNote: Bundled pjproject has been upgraded to 2.14. For more + information on what all is included in this change, check out the + pjproject Github page: https://github.com/pjsip/pjproject/releases

+

app_speech_utils.c: Allow partial speech results.

+

Author: cmaj + Date: 2024-02-02

+

Adds 'p' option to SpeechBackground() application. + With this option, when the app timeout is reached, + whatever the backend speech engine collected will + be returned as if it were the final, full result. + (This works for engines that make partial results.)

+

Resolves: #572

+

UserNote: The SpeechBackground dialplan application now supports a 'p' + option that will return partial results from speech engines that + provide them when a timeout occurs.

+

res_pjsip_outbound_registration.c: Add User-Agent header override

+

Author: Flole998 + Date: 2023-12-13

+

This introduces a setting for outbound registrations to override the + global User-Agent header setting.

+

Resolves: #515

+

UserNote: PJSIP outbound registrations now support a per-registration + User-Agent header

+

utils: Make behavior of ast_strsep* match strsep.

+

Author: Joshua C. Colp + Date: 2024-01-31

+

Given the scenario of passing an empty string to the + ast_strsep functions the functions would return NULL + instead of an empty string. This is counter to how + strsep itself works.

+

This change alters the behavior of the functions to + match that of strsep.

+

Fixes: #565

+

app_chanspy: Add 'D' option for dual-channel audio

+

Author: Mike Bradeen + Date: 2024-01-31

+

Adds the 'D' option to app chanspy that causes the input and output + frames of the spied channel to be interleaved in the spy output frame. + This allows the input and output of the spied channel to be decoded + separately by the receiver.

+

If the 'o' option is also set, the 'D' option is ignored as the + audio being spied is inherently one direction.

+

Fixes: #569

+

UserNote: The ChanSpy application now accepts the 'D' option which + will interleave the spied audio within the outgoing frames. The + purpose of this is to allow the audio to be read as a Dual channel + stream with separate incoming and outgoing audio. Setting both the + 'o' option and the 'D' option and results in the 'D' option being + ignored.

+

app_if: Fix next priority calculation.

+

Author: Naveen Albert + Date: 2024-01-28

+

Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed + a branching issue but "overshoots" when calculating + the next priority. This fixes that; accompanying + test suite tests have also been extended.

+

Resolves: #560

+

res_pjsip_t38.c: Permit IPv6 SDP connection addresses.

+

Author: Sean Bright + Date: 2024-01-29

+

The existing code prevented IPv6 addresses from being properly parsed.

+

Fixes #558

+

BuildSystem: Bump autotools versions on OpenBSD.

+

Author: Brad Smith + Date: 2024-01-27

+

Bump up to the more commonly used and modern versions of + autoconf and automake.

+

main/utils: Simplify the FreeBSD ast_get_tid() handling

+

Author: Brad Smith + Date: 2024-01-27

+

FreeBSD has had kernel threads for 20+ years.

+

res_pjsip_session.c: Correctly format SDP connection addresses.

+

Author: Sean Bright + Date: 2024-01-27

+

Resolves a regression identified by @justinludwig involving the + rendering of IPv6 addresses in outgoing SDP.

+

Also updates media_address on PJSIP endpoints so that if we are able + to parse the configured value as an IP we store it in a format that we + can directly use later. Based on my reading of the code it appeared + that one could configure media_address as:

+

[foo] + type = endpoint + ... + media_address = [2001:db8::]

+

And that value would be blindly copied into the outgoing SDP without + regard to its format.

+

Fixes #541

+

rtp_engine.c: Correct sample rate typo for L16/44100.

+

Author: Sean Bright + Date: 2024-01-28

+

Fixes #555

+

manager.c: Fix erroneous reloads in UpdateConfig.

+

Author: Naveen Albert + Date: 2024-01-25

+

Currently, a reload will always occur if the + Reload header is provided for the UpdateConfig + action. However, we should not be doing a reload + if the header value has a falsy value, per the + documentation, so this makes the reload behavior + consistent with the existing documentation.

+

Resolves: #551

+

res_calendar_icalendar: Print iCalendar error on parsing failure.

+

Author: Naveen Albert + Date: 2023-12-14

+

If libical fails to parse a calendar, print the error message it provdes.

+

Resolves: #492

+

app_confbridge: Don't emit warnings on valid configurations.

+

Author: Sean Bright + Date: 2024-01-21

+

The numeric bridge profile options internal_sample_rate and + maximum_sample_rate are documented to accept the special values + auto and none, respectively. While these values currently work, + they also emit warnings when used which could be confusing for users.

+

In passing, also ensure that we only accept the documented range of + sample rate values between 8000 and 192000.

+

Fixes #546

+

app_voicemail_odbc: remove macrocontext from voicemail_messages table

+

Author: Mike Bradeen + Date: 2024-01-10

+

When app_macro was deprecated, the macrocontext column was removed from + the INSERT statement but the binds were not renumbered. This broke the + insert.

+

This change removes the macrocontext column via alembic and re-numbers + the existing columns in the INSERT.

+

Fixes: #527

+

UserNote: The fix requires removing the macrocontext column from the + voicemail_messages table in the voicemail database via alembic upgrade.

+

UpgradeNote: The fix requires that the voicemail database be upgraded via + alembic. Upgrading to the latest voicemail database via alembic will + remove the macrocontext column from the voicemail_messages table.

+

chan_dahdi: Allow MWI to be manually toggled on channels.

+

Author: Naveen Albert + Date: 2023-11-10

+

This adds a CLI command to manually toggle the MWI status + of a channel, useful for troubleshooting or resetting + MWI devices, similar to the capabilities offered with + SIP messaging to manually control MWI status.

+

UserNote: The 'dahdi set mwi' now allows MWI on channels + to be manually toggled if needed for troubleshooting.

+

Resolves: #440

+

logger: Fix linking regression.

+

Author: Naveen Albert + Date: 2024-01-16

+

Commit 008731b0a4b96c4e6c340fff738cc12364985b64 + caused a regression by resulting in logger.xml + being compiled and linked into the asterisk + binary in lieu of logger.c on certain platforms + if Asterisk was compiled in dev mode.

+

To fix this, we ensure the file has a unique + name without the extension. Most existing .xml + files have been named differently from any + .c files in the same directory or did not + pose this issue.

+

channels/pjsip/dialplan_functions.xml does not + pose this issue but is also being renamed + to adhere to this policy.

+

Resolves: #539

+

chan_rtp.c: MulticastRTP missing refcount without codec option

+

Author: PeterHolik + Date: 2024-01-15

+

Fixes: #529

+

chan_rtp.c: Change MulticastRTP nameing to avoid memory leak

+

Author: PeterHolik + Date: 2024-01-16

+

Fixes: asterisk#536

+

func_frame_trace: Add CLI command to dump frame queue.

+

Author: Naveen Albert + Date: 2024-01-12

+

This adds a simple CLI command that can be used for + analyzing all frames currently queued to a channel.

+

A couple log messages are also adjusted to be more + useful in tracing bridging problems.

+

Resolves: #533

+

menuselect: Use more specific error message.

+

Author: Naveen Albert + Date: 2024-01-04

+

Instead of using the same error message for + missing dependencies and conflicts, be specific + about what actually went wrong.

+

Resolves: #520

+

res_pjsip_nat: Fix potential use of uninitialized transport details

+

Author: Maximilian Fridrich + Date: 2024-01-08

+

The ast_sip_request_transport_details must be zero initialized, + otherwise this could lead to a SEGV.

+

Resolves: #509

+

app_if: Fix faulty EndIf branching.

+

Author: Naveen Albert + Date: 2023-12-23

+

This fixes faulty branching logic for the + EndIf application. Instead of computing + the next priority, which should be done + for false conditionals or ExitIf, we should + simply advance to the next priority.

+

Resolves: #341

+

manager.c: Fix regression due to using wrong free function.

+

Author: Naveen Albert + Date: 2023-12-26

+

Commit 424be345639d75c6cb7d0bd2da5f0f407dbd0bd5 introduced + a regression by calling ast_free on memory allocated by + realpath. This causes Asterisk to abort when executing this + function. Since the memory is allocated by glibc, it should + be freed using ast_std_free.

+

Resolves: #513

+

res_rtp_asterisk: Fix regression issues with DTLS client check

+

Author: George Joseph + Date: 2023-12-15

+ +

Resolves: #500 + Resolves: #503 + Resolves: #505

+

res_pjsip_header_funcs: Duplicate new header value, don't copy.

+

Author: Gitea + Date: 2023-07-10

+

When updating an existing header the 'update' code incorrectly + just copied the new value into the existing buffer. If the + new value exceeded the available buffer size memory outside + of the buffer would be written into, potentially causing + a crash.

+

This change makes it so that the 'update' now duplicates + the new header value instead of copying it into the existing + buffer.

+

res_pjsip: disable raw bad packet logging

+

Author: Mike Bradeen + Date: 2023-07-25

+

Add patch to split the log level for invalid packets received on the + signaling port. The warning regarding the packet will move to level 2 + so that it can still be displayed, while the raw packet will be at level + 4.

+

res_rtp_asterisk.c: Check DTLS packets against ICE candidate list

+

Author: George Joseph + Date: 2023-11-09

+

When ICE is in use, we can prevent a possible DOS attack by allowing + DTLS protocol messages (client hello, etc) only from sources that + are in the active remote candidates list.

+

Resolves: GHSA-hxj9-xwr8-w8pq

+

manager.c: Prevent path traversal with GetConfig.

+

Author: Ben Ford + Date: 2023-11-13

+

When using AMI GetConfig, it was possible to access files outside of the + Asterisk configuration directory by using filenames with ".." and "./" + even while live_dangerously was not enabled. This change resolves the + full path and ensures we are still in the configuration directory before + attempting to access the file.

+

config_options.c: Fix truncation of option descriptions.

+

Author: Naveen Albert + Date: 2023-11-09

+

This increases the format width of option descriptions + to avoid needless truncation for longer descriptions.

+

Resolves: #428

+

manager.c: Improve clarity of "manager show connected".

+

Author: Naveen Albert + Date: 2023-12-05

+

Improve the "manager show connected" CLI command + to clarify that the last two columns are permissions + related, not counts, and use sufficient widths + to consistently display these values.

+

ASTERISK-30143 #close + Resolves: #482

+

make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.

+

Author: Sean Bright + Date: 2023-12-01

+

Although make_xml_documentation's print_dependencies command was + corrected by the previous fix (#461) for #142, the create_xml was + not properly handling LOCAL_MOD_SUBDIRS XML documentation.

+

general: Fix broken links.

+

Author: Naveen Albert + Date: 2023-11-09

+

This fixes a number of broken links throughout the + tree, mostly caused by wiki.asterisk.org being replaced + with docs.asterisk.org, which should eliminate the + need for sporadic fixes as in f28047db36a70e81fe373a3d19132c43adf3f74b.

+

Resolves: #430

+

MergeApproved.yml: Remove unneeded concurrency

+

Author: George Joseph + Date: 2023-12-06

+

The concurrency parameter on the MergeAndCherryPick job has + been rmeoved. It was a hold-over from earlier days.

+

app_dial: Add option "j" to preserve initial stream topology of caller

+

Author: Maximilian Fridrich + Date: 2023-11-30

+

Resolves: #462

+

UserNote: The option "j" is now available for the Dial application which + uses the initial stream topology of the caller to create the outgoing + channels.

+

pbx_config.c: Don't crash when unloading module.

+

Author: Sean Bright + Date: 2023-12-02

+

pbx_config subscribes to manager events to capture the FullyBooted + event but fails to unsubscribe if the module is loaded after that + event fires. If the module is unloaded, a crash occurs the next time a + manager event is raised.

+

We now unsubscribe when the module is unloaded if we haven't already + unsubscribed.

+

Fixes #470

+

ast_coredumper: Increase reliability

+

Author: George Joseph + Date: 2023-11-11

+

Instead of searching for the asterisk binary and the modules in the + filesystem, we now get their locations, along with libdir, from + the coredump itself...

+

For the binary, we can use gdb -c <coredump> ... "info proc exe". + gdb can print this even without having the executable and symbols.

+

Once we have the binary, we can get the location of the modules with + gdb ... "print ast_config_AST_MODULE_DIR

+

If there was no result then either it's not an asterisk coredump + or there were no symbols loaded. Either way, it's not usable.

+

For libdir, we now run "strings" on the note0 section of the + coredump (which has the shared library -> memory address xref) and + search for "libasteriskssl|libasteriskpj", then take the dirname.

+

Since we're now getting everything from the coredump, it has to be + correct as long as we're not crossing namespace boundaries like + running asterisk in a docker container but trying to run + ast_coredumper from the host using a shared file system (which you + shouldn't be doing).

+

There is still a case for using --asterisk-bin and/or --libdir: If + you've updated asterisk since the coredump was taken, the binary, + libraries and modules won't match the coredump which will render it + useless. If you can restore or rebuild the original files that + match the coredump and place them in a temporary directory, you can + use --asterisk-bin, --libdir, and a new --moddir option to point to + them and they'll be correctly captured in a tarball created + with --tarball-coredumps. If you also use --tarball-config, you can + use a new --etcdir option to point to what normally would be the + /etc/asterisk directory.

+

Also addressed many "shellcheck" findings.

+

Resolves: #445

+

logger.c: Move LOG_GROUP documentation to dedicated XML file.

+

Author: Sean Bright + Date: 2023-12-01

+

The get_documentation awk script will only extract the first + DOCUMENTATION block that it finds in a given file. This is by design + (9bc2127) to prevent AMI event documentation from being pulled in to + the core.xml documentation file.

+

Because of this, the LOG_GROUP documentation added in 89709e2 was + not being properly extracted and was missing fom the resulting XML + documentation file. This commit moves the LOG_GROUP documentation to + a separate logger.xml file.

+

res_odbc.c: Allow concurrent access to request odbc connections

+

Author: Matthew Fredrickson + Date: 2023-11-30

+

There are valid scenarios where res_odbc's connection pool might have some dead + or stuck connections while others are healthy (imagine network + elements/firewalls/routers silently timing out connections to a single DB and a + single IP address, or a heterogeneous connection pool connected to potentially + multiple IPs/instances of a replicated DB using a DNS front end for load + balancing and one replica fails).

+

In order to time out those unhealthy connections without blocking access to + other parts of Asterisk that may attempt access to the connection pool, it would + be beneficial to not lock/block access around the entire pool in + _ast_odbc_request_obj2 while doing potentially blocking operations on connection + pool objects such as the connection_dead() test, odbc_obj_connect(), or by + dereferencing a struct odbc_obj for the last time and triggering a + odbc_obj_disconnect().

+

This would facilitate much quicker and concurrent timeout of dead connections + via the connection_dead() test, which could block potentially for a long period + of time depending on odbc.ini or other odbc connector specific timeout settings.

+

This also would make rapid failover (in the clustered DB scenario) much quicker.

+

This patch changes the locking in _ast_odbc_request_obj2() to not lock around + odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to + lock around truly shared, non-immutable state like the connection_cnt member and + the connections list on struct odbc_class.

+

Fixes: #465

+

res_pjsip_header_funcs.c: Check URI parameter length before copying.

+

Author: Sean Bright + Date: 2023-12-04

+

Fixes #477

+

config.c: Log #exec include failures.

+

Author: Sean Bright + Date: 2023-11-22

+

If the script referenced by #exec does not exist, writes anything to + stderr, or exits abnormally or with a non-zero exit status, we log + that to Asterisk's error logging channel.

+

Additionally, write out a warning if the script produces no output.

+

Fixes #259

+

make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.

+

Author: Sean Bright + Date: 2023-11-27

+

If LOCAL_MOD_SUBDIRS contains absolute paths, do not prefix them with + the path to Asterisk's source tree.

+

Fixes #142

+

app_voicemail.c: Completely resequence mailbox folders.

+

Author: Sean Bright + Date: 2023-11-27

+

Resequencing is a process that occurs when we open a voicemail folder + and discover that there are gaps between messages (e.g. msg0000.txt + is missing but msg0001.txt exists). Resequencing involves shifting + the existing messages down so we end up with a sequential list of + messages.

+

Currently, this process stops after reaching a threshold based on the + message limit (maxmsg) configured on the current folder. However, if + maxmsg is lowered when a voicemail folder contains more than + maxmsg + 10 messages, resequencing will not run completely leaving + the mailbox in an inconsistent state.

+

We now resequence up to the maximum number of messages permitted by + app_voicemail (currently hard-coded at 9999 messages).

+

Fixes #86

+

sig_analog: Fix channel leak when mwimonitor is enabled.

+

Author: Naveen Albert + Date: 2023-11-24

+

When mwimonitor=yes is enabled for an FXO port, + the do_monitor thread will launch mwi_thread if it thinks + there could be MWI on an FXO channel, due to the noise + threshold being satisfied. This, in turns, calls + analog_ss_thread_start in sig_analog. However, unlike + all other instances where __analog_ss_thread is called + in sig_analog, this call path does not properly set + pvt->ss_astchan to the Asterisk channel, which means + that the Asterisk channel is NULL when __analog_ss_thread + starts executing. As a result, the thread exits and the + channel is never properly cleaned up by calling ast_hangup.

+

This caused issues with do_monitor on incoming calls, + as it would think the channel was still owned even while + receiving events, leading to an infinite barrage of + warning messages; additionally, the channel would persist + improperly.

+

To fix this, the assignment is added to the call path + where it is missing (which is only used for mwi_thread). + A warning message is also added since previously there + was no indication that __analog_ss_thread was exiting + abnormally. This resolves both the channel leak and the + condition that led to the warning messages.

+

Resolves: #458

+

res_rtp_asterisk.c: Update for OpenSSL 3+.

+

Author: Sean Bright + Date: 2023-11-20

+

In 5ac5c2b0 we defined OPENSSL_SUPPRESS_DEPRECATED to silence + deprecation warnings. This commit switches over to using + non-deprecated API.

+

alembic: Update list of TLS methods available on ps_transports.

+

Author: Sean Bright + Date: 2023-11-14

+

Related to #221 and #222.

+

Also adds *.ini to the .gitignore file in ast-db-manage for + convenience.

+

func_channel: Expose previously unsettable options.

+

Author: Naveen Albert + Date: 2023-11-11

+

Certain channel options are not set anywhere or + exposed in any way to users, making them unusable. + This exposes some of these options which make sense + for users to manipulate at runtime.

+

Resolves: #442

+

app.c: Allow ampersands in playback lists to be escaped.

+

Author: Sean Bright + Date: 2023-11-07

+

Any function or application that accepts a &-separated list of + filenames can now include a literal & in a filename by wrapping the + entire filename in single quotes, e.g.:

+

exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world)

+

Fixes #172

+

UpgradeNote: Ampersands in URLs passed to the Playback(), + Background(), SpeechBackground(), Read(), Authenticate(), or + Queue() applications as filename arguments can now be escaped by + single quoting the filename. Additionally, this is also possible when + using the CONFBRIDGE dialplan function, or configuring various + features in confbridge.conf and queues.conf.

+

uri.c: Simplify ast_uri_make_host_with_port()

+

Author: Sean Bright + Date: 2023-11-09

+

func_curl.c: Remove CURLOPT() plaintext documentation.

+

Author: Sean Bright + Date: 2023-11-13

+

I assume this was missed when initially converting to XML + documentation and we've been kicking the can down the road since.

+

res_http_websocket.c: Set hostname on client for certificate validation.

+

Author: Sean Bright + Date: 2023-11-09

+

Additionally add a assert() to in the TLS client setup code to + ensure that hostname is set when it is supposed to be.

+

Fixes #433

+

live_ast: Add astcachedir to generated asterisk.conf.

+

Author: Sean Bright + Date: 2023-11-09

+

astcachedir (added in b0842713) was not added to live_ast so + continued to point to the system /var/cache directory instead of the + one in the live environment.

+

SECURITY.md: Update with correct documentation URL

+

Author: George Joseph + Date: 2023-11-09

+

func_lock: Add missing see-also refs to documentation.

+

Author: Naveen Albert + Date: 2023-11-09

+

Resolves: #423

+

app_followme.c: Grab reference on nativeformats before using it

+

Author: Matthew Fredrickson + Date: 2023-10-25

+

Fixes a crash due to a lack of proper reference on the nativeformats + object before passing it into ast_request(). Also found potentially + similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c

+

Fixes: #388

+

configs: Improve documentation for bandwidth in iax.conf.

+

Author: Naveen Albert + Date: 2023-11-09

+

This improves the documentation for the bandwidth setting + in iax.conf by making it clearer what the ramifications + of this setting are. It also changes the sample default + from low to high, since only high is compatible with good + codecs that people will want to use in the vast majority + of cases, and this is a common gotcha that trips up new users.

+

Resolves: #425

+

logger: Add channel-based filtering.

+

Author: Naveen Albert + Date: 2023-08-09

+

This adds the ability to filter console + logging by channel or groups of channels. + This can be useful on busy systems where + an administrator would like to analyze certain + calls in detail. A dialplan function is also + included for the purpose of assigning a channel + to a group (e.g. by tenant, or some other metric).

+

ASTERISK-30483 #close

+

Resolves: #242

+

UserNote: The console log can now be filtered by + channels or groups of channels, using the + logger filter CLI commands.

+

chan_iax2.c: Don't send unsanitized data to the logger.

+

Author: Sean Bright + Date: 2023-11-08

+

This resolves an issue where non-printable characters could be sent to + the console/log files.

+

codec_ilbc: Disable system ilbc if version >= 3.0.0

+

Author: George Joseph + Date: 2023-11-07

+

Fedora 37 started shipping ilbc 3.0.4 which we don't yet support. + configure.ac now checks the system for "libilbc < 3" instead of + just "libilbc". If true, the system version of ilbc will be used. + If not, the version included at codecs/ilbc will be used.

+

Resolves: #84

+

resource_channels.c: Explicit codec request when creating UnicastRTP.

+

Author: Sean Bright + Date: 2023-11-06

+

Fixes #394

+

doc: Update IP Quality of Service links.

+

Author: Sean Bright + Date: 2023-11-07

+

Fixes #328

+

chan_pjsip: Add PJSIPHangup dialplan app and manager action

+

Author: George Joseph + Date: 2023-10-31

+

See UserNote below.

+

Exposed the existing Hangup AMI action in manager.c so we can use + all of it's channel search and AMI protocol handling without + duplicating that code in dialplan_functions.c.

+

Added a lookup function to res_pjsip.c that takes in the + string represenation of the pjsip_status_code enum and returns + the actual status code. I.E. ast_sip_str2rc("DECLINE") returns + 603. This allows the caller to specify PJSIPHangup(decline) in + the dialplan, just like Hangup(call_rejected).

+

Also extracted the XML documentation to its own file since it was + almost as large as the code itself.

+

UserNote: A new dialplan app PJSIPHangup and AMI action allows you + to hang up an unanswered incoming PJSIP call with a specific SIP + response code in the 400 -> 699 range.

+

chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.

+

Author: Sean Bright + Date: 2023-11-06

+

When IAX2 debugging was enabled (iax2 set debug on), if the last IE + in a frame was one that may not have any data - such as the CALLTOKEN + IE in an NEW request - it was not getting displayed.

+

chan_dahdi: Warn if nonexistent cadence is requested.

+

Author: Naveen Albert + Date: 2023-11-02

+

If attempting to ring a channel using a nonexistent cadence, + emit a warning, before falling back to the default cadence.

+

Resolves: #409

+

stasis: Update the snapshot after setting the redirect

+

Author: Holger Hans Peter Freyther + Date: 2023-10-21

+

The previous commit added the caller_rdnis attribute. Make it + avialble during a possible ChanngelHangupRequest.

+

ari: Provide the caller ID RDNIS for the channels

+

Author: Holger Hans Peter Freyther + Date: 2023-10-14

+

Provide the caller ID RDNIS when available. This will allow an + application to follow the redirect.

+

main/utils: Implement ast_get_tid() for OpenBSD

+

Author: Brad Smith + Date: 2023-11-01

+

Implement the ast_get_tid() function for OpenBSD. OpenBSD supports + getting the TID via getthrid().

+

res_rtp_asterisk.c: Fix runtime issue with LibreSSL

+

Author: Brad Smith + Date: 2023-11-02

+

The module will fail to load. Use proper function DTLS_method() with LibreSSL.

+

app_directory: Add ADSI support to Directory.

+

Author: Naveen Albert + Date: 2023-09-27

+

This adds optional ADSI support to the Directory + application, which allows callers with ADSI CPE + to navigate the Directory system significantly + faster than is possible using the audio prompts. + Callers can see the directory name (and optionally + extension) on their screenphone and confirm or + reject a match immediately rather than waiting + for it to be spelled out, enhancing usability.

+

Resolves: #356

+

core_local: Fix local channel parsing with slashes.

+

Author: Naveen Albert + Date: 2023-08-09

+

Currently, trying to call a Local channel with a slash + in the extension will fail due to the parsing of characters + after such a slash as being dial modifiers. Additionally, + core_local is inconsistent and incomplete with + its parsing of Local dial strings in that sometimes it + uses the first slash and at other times it uses the last.

+

For instance, something like DAHDI/5 or PJSIP/device + is a perfectly usable extension in the dialplan, but Local + channels in particular prevent these from being called.

+

This creates inconsistent behavior for users, since using + a slash in an extension is perfectly acceptable, and using + a Goto to accomplish this works fine, but if specified + through a Local channel, the parsing prevents this.

+

This fixes this by explicitly parsing options from the + last slash in the extension, rather than the first one, + which doesn't cause an issue for extensions with slashes.

+

ASTERISK-30013 #close

+

Resolves: #248

+

Remove files that are no longer updated

+

Author: Mark Murawski + Date: 2023-10-30

+

Fixes: #360

+

app_voicemail: Add AMI event for mailbox PIN changes.

+

Author: Naveen Albert + Date: 2023-10-30

+

This adds an AMI event that is emitted whenever a + mailbox password is successfully changed, allowing + AMI consumers to process these.

+

UserNote: The VoicemailPasswordChange event is + now emitted whenever a mailbox password is updated, + containing the mailbox information and the new + password.

+

Resolves: #398

+

app_queue.c: Emit unpause reason with PauseQueueMember event.

+

Author: Sean Bright + Date: 2023-10-30

+

Fixes #395

+

bridge_simple: Suppress unchanged topology change requests

+

Author: George Joseph + Date: 2023-10-30

+

In simple_bridge_join, we were sending topology change requests + even when the new and old topologies were the same. In some + circumstances, this can cause unnecessary re-invites and even + a re-invite flood. We now suppress those.

+

Resolves: #384

+

res_pjsip: Include cipher limit in config error message.

+

Author: Naveen Albert + Date: 2023-10-30

+

If too many ciphers are specified in the PJSIP config, + include the maximum number of ciphers that may be + specified in the user-facing error message.

+

Resolves: #396

+

res_speech: allow speech to translate input channel

+

Author: Mike Bradeen + Date: 2023-09-07

+ +

Resolves: #129

+

UserNote: res_speech now supports translation of an input channel + to a format supported by the speech provider, provided a translation + path is available between the source format and provider capabilites.

+

res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.

+

Author: Sean Bright + Date: 2023-10-25

+

Fixes #386

+

res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.

+

Author: Sean Bright + Date: 2023-10-17

+

Fixes #376

+

Update issue guidelines link for bug reports.

+

Author: Joshua C. Colp + Date: 2023-10-27

+

api.wiki.mustache: Fix indentation in generated markdown

+

Author: George Joseph + Date: 2023-10-25

+

The '*' list indicator for default values and allowable values for + path, query and POST parameters need to be indented 4 spaces + instead of 2.

+

Should resolve issue 38 in the documentation repo.

+

pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

+

Author: Sean Bright + Date: 2023-10-23

+

Per RFC8827:

+
  Implementations MUST NOT implement DTLS renegotiation and MUST
+  reject it with a "no_renegotiation" alert if offered.
+
+

So we disable it when webrtc=yes is set.

+

Fixes #378

+

UpgradeNote: The dtls_rekey will be disabled if webrtc support is + requested on an endpoint. A warning will also be emitted.

+

configs: Fix typo in pjsip.conf.sample.

+

Author: Samuel Olaechea + Date: 2023-10-12

+

res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown

+

Author: George Joseph + Date: 2023-10-19

+

Commit f66f77f last year prevents the res_pjsip_exten_state and + res_pjsip_mwi modules from unloading due to possible pjproject + asserts if the modules are reloaded. A side effect of the + implementation is that the taskprocessors these modules use aren't + being released. When asterisk is doing a graceful shutdown, it + waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all + taskprocessors to stop but since those 2 modules don't release + theirs, the shutdown hangs for that amount of time.

+

This change allows the modules to be unloaded and their resources to + be released when ast_shutdown_final is true.

+

Resolves: #379

+

res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters

+

Author: sungtae kim + Date: 2023-09-23

+

This commit introduces an extension to the endpoint and relevant + resource sizes for PJSIP, transitioning from its current 40-character + constraint to a more versatile 255-character capacity. This enhancement + significantly overcomes limitations related to domain qualification and + practical usage, ultimately delivering improved functionality. In + addition, it includes adjustments to accommodate the expanded realm size + within the ARI, specifically enhancing the maximum realm length.

+

Resolves: #345

+

UserNote: With this update, the PJSIP realm lengths have been extended + to support up to 255 characters.

+

UpgradeNote: As part of this update, the maximum allowable length + for PJSIP endpoints and relevant resources has been increased from + 40 to 255 characters. To take advantage of this enhancement, it is + recommended to run the necessary procedures (e.g., Alembic) to + update your schemas.

+

res_stasis: signal when new command is queued

+

Author: Mike Bradeen + Date: 2023-10-02

+

res_statsis's app loop sleeps for up to .2s waiting on input + to a channel before re-checking the command queue. This can + cause delays between channel setup and bridge.

+

This change is to send a SIGURG on the sleeping thread when + a new command is enqueued. This exits the sleeping thread out + of the ast_waitfor() call triggering the new command being + processed on the channel immediately.

+

Resolves: #362

+

UserNote: Call setup times should be significantly improved + when using ARI.

+

ari/stasis: Indicate progress before playback on a bridge

+

Author: Holger Hans Peter Freyther + Date: 2023-10-02

+

Make it possible to start a playback and the calling party + to receive audio on a bridge before the call is connected.

+

Model the implementation after play_on_channel and deliver a + AST_CONTROL_PROGRESS before starting the playback.

+

For a PJSIP channel this will result in sending a SIP 183 + Session Progress.

+

func_curl.c: Ensure channel is locked when manipulating datastores.

+

Author: Sean Bright + Date: 2023-10-09

+

logger.h: Add ability to change the prefix on SCOPE_TRACE output

+

Author: George Joseph + Date: 2023-10-05

+

You can now define the TRACE_PREFIX macro to change the + default trace line prefix of "file:line function" to + something else. Full documentation in logger.h.

+

res_pjsip: update qualify_timeout documentation with DNS note

+

Author: Mike Bradeen + Date: 2023-09-26

+

The documentation on qualify_timeout does not explicitly state that the timeout + includes any time required to perform any needed DNS queries on the endpoint.

+

If the OPTIONS response is delayed due to the DNS query, it can still render an + endpoint as Unreachable if the net time is enough for qualify_timeout to expire.

+

Resolves: #352

+

res_speech_aeap: add aeap error handling

+

Author: Mike Bradeen + Date: 2023-09-21

+

res_speech_aeap previously did not register an error handler + with aeap, so it was not notified of a disconnect. This resulted + in SpeechBackground never exiting upon a websocket disconnect.

+

Resolves: #303

+

Add libjwt to third-party

+

Author: George Joseph + Date: 2023-09-21

+

The current STIR/SHAKEN implementation is not currently usable due + to encryption issues. Rather than trying to futz with OpenSSL and + the the current code, we can take advantage of the existing + capabilities of libjwt but we first need to add it to the + third-party infrastructure already in place for jansson and + pjproject.

+

A few tweaks were also made to the third-party infrastructure as + a whole. The jansson "dest" install directory was renamed "dist" + to better match convention, and the third-party Makefile was updated + to clean all product directories not just the ones currently in + use.

+

Resolves: #349

+

chan_dahdi: Clarify scope of callgroup/pickupgroup.

+

Author: Naveen Albert + Date: 2023-09-04

+

Internally, chan_dahdi only applies callgroup and + pickupgroup to FXO signalled channels, but this is + not documented anywhere. This is now documented in + the sample config, and a warning is emitted if a + user tries configuring these settings for channel + types that do not support these settings, since they + will not have any effect.

+

Resolves: #294

+

func_json: Fix crashes for some types

+

Author: Bastian Triller + Date: 2023-09-21

+

This commit fixes crashes in JSON_DECODE() for types null, true, false + and real numbers.

+

In addition it ensures that a path is not deeper than 32 levels.

+

Also allow root object to be an array.

+

Add unit tests for above cases.

+

app_voicemail: Disable ADSI if unavailable.

+

Author: Naveen Albert + Date: 2023-09-27

+

If ADSI is available on a channel, app_voicemail will repeatedly + try to use ADSI, even if there is no CPE that supports it. This + leads to many unnecessary delays during the session. If ADSI is + available but ADSI setup fails, we now disable it to prevent + further attempts to use ADSI during the session.

+

Resolves: #354

+

codec_builtin: Use multiples of 20 for maximum_ms

+

Author: Eduardo + Date: 2023-07-28

+

Some providers require a multiple of 20 for the maxptime or fail to complete calls, + e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.

+

Resolves: #260

+

lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

+

Author: George Joseph + Date: 2023-09-13

+

Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. + Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead + to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just + causes the lock calls to loop over trylock in 200us intervals until + the lock is obtained and spits out log messages if it takes more + than 5 seconds. From a code perspective, the only reason they were + tied together was for logging. So... The ifdefs in lock.c were + refactored to allow DETECT_DEADLOCKS to be enabled without + also enabling DEBUG_THREADS.

+

Resolves: #321

+

UserNote: You no longer need to select DEBUG_THREADS to use + DETECT_DEADLOCKS. This removes a significant amount of overhead + if you just want to detect possible deadlocks vs needing full + lock tracing.

+

asterisk.c: Use the euid's home directory to read/write cli history

+

Author: George Joseph + Date: 2023-09-15

+

The CLI .asterisk_history file is read from/written to the directory + specified by the HOME environment variable. If the root user starts + asterisk with the -U/-G options, or with runuser/rungroup set in + asterisk.conf, the asterisk process is started as root but then it + calls setuid/setgid to set the new user/group. This does NOT reset + the HOME environment variable to the new user's home directory + though so it's still left as "/root". In this case, the new user + will almost certainly NOT have access to read from or write to the + history file.

+ +

Resolves: #337

+

res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.

+

Author: Tinet-mucw + Date: 2023-09-13

+

From the gdb information, ast_websocket_read reads a message successfully, + then transport_read is called in the serializer. During execution of pjsip_transport_down, + ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop. + After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages. + This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue. + In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.

+

Resolves: asterisk#299

+

cel: add publish user event helper

+

Author: Mike Bradeen + Date: 2023-09-14

+

Add a wrapper function around ast_cel_publish_event that + packs event and extras into a blob before publishing

+

Resolves:#330

+

chan_console: Fix deadlock caused by unclean thread exit.

+

Author: Naveen Albert + Date: 2023-09-09

+

To terminate a console channel, stop_stream causes pthread_cancel + to make stream_monitor exit. However, commit 5b8fea93d106332bc0faa4b7fa8a6ea71e546cac + added locking to this function which results in deadlock due to + the stream_monitor thread being killed while it's holding the pvt lock.

+

To resolve this, a flag is now set and read to indicate abort, so + the use of pthread_cancel and pthread_kill can be avoided altogether.

+

Resolves: #308

+

file.c: Add ability to search custom dir for sounds

+

Author: George Joseph + Date: 2023-09-11

+

To better co-exist with sounds files that may be managed by + packages, custom sound files may now be placed in + AST_DATA_DIR/sounds/custom instead of the standard + AST_DATA_DIR/sounds/ directory. If the new + "sounds_search_custom_dir" option in asterisk.conf is set + to "true", asterisk will search the custom directory for sounds + files before searching the standard directory. For performance + reasons, the "sounds_search_custom_dir" defaults to "false".

+

Resolves: #315

+

UserNote: A new option "sounds_search_custom_dir" has been added to + asterisk.conf that allows asterisk to search + AST_DATA_DIR/sounds/custom for sounds files before searching the + standard AST_DATA_DIR/sounds/ directory.

+

chan_iax2: Improve authentication debugging.

+

Author: Naveen Albert + Date: 2023-08-30

+

Improves and adds some logging to make it easier + for users to debug authentication issues.

+

Resolves: #286

+

res_rtp_asterisk: fix wrong counter management in ioqueue objects

+

Author: Vitezslav Novy + Date: 2023-09-05

+

In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased + which prevents unused ICE TURN threads from being removed.

+

Resolves: #301

+

res_pjsip_pubsub: Add body_type to test_handler for unit tests

+

Author: George Joseph + Date: 2023-09-15

+

The ast_sip_subscription_handler "test_handler" used for the unit + tests didn't set "body_type" so the NULL value was causing + a SEGV in build_subscription_tree(). It's now set to "".

+

Resolves: #335

+

make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

+

Author: George Joseph + Date: 2023-09-13

+

The previous behavior of make_buildopts_h was to not add the + non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, + REF_DEBUG, etc. to the buildopts.h file because "it caused + ccache to invalidate files and extended compile times". They're + only defined by passing them on the gcc command line with '-D' + options. In practice, including them in the include file rarely + causes any impact because the only time ccache cares is if you + actually change an option so the hit occurrs only once after + you change it.

+

OK so why would we want to include them? Many IDEs follow the + include files to resolve defines and if the options aren't in an + include file, it can cause the IDE to mark blocks of "ifdeffed" + code as unused when they're really not.

+

So...

+ +

UserNote: The "Build Options" entry in the "core show settings" + CLI command has been renamed to "ABI related Build Options" and + a new entry named "All Build Options" has been added that shows + both breaking and non-breaking options.

+

func_periodic_hook: Add hangup step to avoid timeout

+

Author: Mike Bradeen + Date: 2023-09-12

+

func_periodic_hook does not hangup after playback, relying on hangup + which keeps the channel alive longer than necessary.

+

Resolves: #325

+

res_stasis_recording.c: Save recording state when unmuted.

+

Author: Sean Bright + Date: 2023-09-12

+

Fixes #322

+

res_speech_aeap: check for null format on response

+

Author: Mike Bradeen + Date: 2023-09-08

+ +

func_periodic_hook: Don't truncate channel name

+

Author: George Joseph + Date: 2023-09-11

+

func_periodic_hook was truncating long channel names which + causes issues when you need to run other dialplan functions/apps + on the channel.

+

Resolves: #319

+

safe_asterisk: Change directory permissions to 755

+

Author: George Joseph + Date: 2023-09-11

+

If the safe_asterisk script detects that the /var/lib/asterisk + directory doesn't exist, it now creates it with 755 permissions + instead of 770. safe_asterisk needing to create that directory + should be extremely rare though because it's normally created + by 'make install' which already sets the permissions to 755.

+

Resolves: #316

+

chan_rtp: Implement RTP glue for UnicastRTP channels

+

Author: Maximilian Fridrich + Date: 2023-09-05

+

Resolves: #298

+

UserNote: The dial string option 'g' was added to the UnicastRTP channel + which enables RTP glue and therefore native RTP bridges with those + channels.

+

variables: Add additional variable dialplan functions.

+

Author: Joshua C. Colp + Date: 2023-08-31

+

Using the Set dialplan application does not actually + delete channel or global variables. Instead the + variables are set to an empty value.

+

This change adds two dialplan functions, + GLOBAL_DELETE and DELETE which can be used to + delete global and channel variables instead + of just setting them to empty.

+

There is also no ability within the dialplan to + determine if a global or channel variable has + actually been set or not.

+

This change also adds two dialplan functions, + GLOBAL_EXISTS and VARIABLE_EXISTS which can be + used to determine if a global or channel variable + has been set or not.

+

Resolves: #289

+

UserNote: Four new dialplan functions have been added. + GLOBAL_DELETE and DELETE have been added which allows + the deletion of global and channel variables. + GLOBAL_EXISTS and VARIABLE_EXISTS have been added + which checks whether a global or channel variable has + been set.

+

ari-stubs: Fix more local anchor references

+

Author: George Joseph + Date: 2023-09-05

+

Also allow CreateDocs job to be run manually with default branches.

+

ari-stubs: Fix broken documentation anchors

+

Author: George Joseph + Date: 2023-09-05

+

All of the links that reference page anchors with capital letters in + the ids (#Something) have been changed to lower case to match the + anchors that are generated by mkdocs.

+

res_pjsip_session: Send Session Interval too small response

+

Author: Bastian Triller + Date: 2023-08-28

+

Handle session interval lower than endpoint's configured minimum timer + when sending first answer. Timer setting is checked during this step and + needs to handled appropriately. + Before this change, no response was sent at all. After this change a + response with 422 Session Interval too small is sent to UAC.

+

app_dial: Fix infinite loop when sending digits.

+

Author: Naveen Albert + Date: 2023-08-28

+

If the called party hangs up while digits are being + sent, -1 is returned to indicate so, but app_dial + was not checking the return value, resulting in + the hangup being lost and looping forever until + the caller manually hangs up the channel. We now + abort if digit sending fails.

+

ASTERISK-29428 #close

+

Resolves: #281

+

app_voicemail: Fix for loop declarations

+

Author: Mike Bradeen + Date: 2023-08-29

+

Resolve for loop initial declarations added in cli changes.

+

Resolves: #275

+

alembic: Fix quoting of the 100rel column

+

Author: George Joseph + Date: 2023-08-28

+

Add quoting around the ps_endpoints 100rel column in the ALTER + statements. Although alembic doesn't complain when generating + sql statements, postgresql does (rightly so).

+

Resolves: #274

+

pbx.c: Fix gcc 12 compiler warning.

+

Author: Naveen Albert + Date: 2023-08-27

+

Resolves: #277

+

app_audiosocket: Fixed timeout with -1 to avoid busy loop.

+

Author: zhengsh + Date: 2023-08-24

+

Resolves: asterisk#234

+

download_externals: Fix a few version related issues

+

Author: George Joseph + Date: 2023-08-18

+ +

Resolves: #263

+

main/refer.c: Fix double free in refer_data_destructor + potential leak

+

Author: Maximilian Fridrich + Date: 2023-08-21

+

Resolves: #267

+

sig_analog: Add Called Subscriber Held capability.

+

Author: Naveen Albert + Date: 2023-08-09

+

This adds support for Called Subscriber Held for FXS + lines, which allows users to go on hook when receiving + a call and resume the call later from another phone on + the same line, without disconnecting the call. This is + a convenience mechanism that most real PSTN telephone + switches support.

+

ASTERISK-30372 #close

+

Resolves: #240

+

UserNote: Called Subscriber Held is now supported for analog + FXS channels, using the calledsubscriberheld option. This allows + a station user to go on hook when receiving an incoming call + and resume from another phone on the same line by going on hook, + without disconnecting the call.

+

install_prereq: Fix dependency install on aarch64.

+

Author: Jason D. McCormick + Date: 2023-04-28

+

Fixes dependency solutions in install_prereq for Debian aarch64 + platforms. install_prereq was attempting to forcibly install 32-bit + armhf packages due to the aptitude search for dependencies.

+

Resolves: #37

+

res_pjsip.c: Set contact_user on incoming call local Contact header

+

Author: MikeNaso + Date: 2023-08-08

+

If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.

+

Resolves: #226

+

extconfig: Allow explicit DB result set ordering to be disabled.

+

Author: Sean Bright + Date: 2023-07-12

+

Added a new boolean configuration flag - + order_multi_row_results_by_initial_column - to both res_pgsql.conf + and res_config_odbc.conf that allows the administrator to disable the + explicit ORDER BY that was previously being added to all generated + SQL statements that returned multiple rows.

+

Fixes: #179

+

res_pjsip_header_funcs: Make prefix argument optional.

+

Author: Naveen Albert + Date: 2023-08-09

+

The documentation for PJSIP_HEADERS claims that + prefix is optional, but in the code it is actually not. + However, there is no inherent reason for this, as users + may want to retrieve all header names, not just those + beginning with a certain prefix.

+

This makes the prefix optional for this function, + simply fetching all header names if not specified. + As a result, the documentation is now correct.

+

Resolves: #230

+

UserNote: The prefix argument to PJSIP_HEADERS is now + optional. If not specified, all header names will be + returned.

+

pjproject_bundled: Increase PJSIP_MAX_MODULE to 38

+

Author: George Joseph + Date: 2023-08-11

+

The default is 32 with 8 being used by pjproject itself. Recent + commits have put us over the limit resulting in assertions in + pjproject. Since this value is used in invites, dialogs, + transports and subscriptions as well as the global pjproject + endpoint, we don't want to increase it too much.

+

Resolves: #255

+

manager: Tolerate stasis messages with no channel snapshot.

+

Author: Joshua C. Colp + Date: 2023-08-09

+

In some cases I have yet to determine some stasis messages may + be created without a channel snapshot. This change adds some + tolerance to this scenario, preventing a crash from occurring.

+

Prepare master for Asterisk 22

+

Author: George Joseph + Date: 2023-08-09

+

core/ari/pjsip: Add refer mechanism

+

Author: Maximilian Fridrich + Date: 2023-05-10

+

This change adds support for refers that are not session based. It + includes a refer implementation for the PJSIP technology which results + in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be + triggered using the new ARI endpoint /endpoints/refer.

+

Resolves: #71

+

UserNote: There is a new ARI endpoint /endpoints/refer for referring + an endpoint to some URI or endpoint.

+

chan_dahdi: Allow autoreoriginating after hangup.

+

Author: Naveen Albert + Date: 2023-08-04

+

Currently, if an FXS channel is still off hook when + all calls on the line have hung up, the user is provided + reorder tone until going back on hook again.

+

In addition to not reflecting what most commercial switches + actually do, it's very common for switches to automatically + reoriginate for the user so that dial tone is provided without + the user having to depress and release the hookswitch manually. + This can increase convenience for users.

+

This behavior is now supported for kewlstart FXS channels. + It's supported only for kewlstart (FXOKS) mainly because the + behavior doesn't make any sense for ground start channels, + and loop start signalling doesn't provide the necessary DAHDI + event that makes this easy to implement. Likely almost everyone + is using FXOKS over FXOLS anyways since FXOLS is pretty useless + these days.

+

ASTERISK-30357 #close

+

Resolves: #224

+

UserNote: The autoreoriginate setting now allows for kewlstart FXS + channels to automatically reoriginate and provide dial tone to the + user again after all calls on the line have cleared. This saves users + from having to manually hang up and pick up the receiver again before + making another call.

+

audiohook: Unlock channel in mute if no audiohooks present.

+

Author: Joshua C. Colp + Date: 2023-08-09

+

In the case where mute was called on a channel that had no + audiohooks the code was not unlocking the channel, resulting + in a deadlock.

+

Resolves: #233

+

sig_analog: Allow three-way flash to time out to silence.

+

Author: Naveen Albert + Date: 2023-07-10

+

sig_analog allows users to flash and use the three-way dial + tone as a primitive hold function, simply by never timing + it out.

+

Some systems allow this dial tone to time out to silence, + so the user is not annoyed by a persistent dial tone. + This option allows the dial tone to time out normally to + silence.

+

ASTERISK-30004 #close + Resolves: #205

+

UserNote: The threewaysilenthold option now allows the three-way + dial tone to time out to silence, rather than continuing forever.

+

res_prometheus: Do not generate broken metrics

+

Author: Holger Hans Peter Freyther + Date: 2023-04-07

+

In 8d6fdf9c3adede201f0ef026dab201b3a37b26b6 invisible bridges were + skipped but that lead to producing metrics with no name and no help.

+

Keep track of the number of metrics configured and then only emit these. + Add a basic testcase that verifies that there is no '(NULL)' in the + output.

+

ASTERISK-30474

+

res_pjsip: Enable TLS v1.3 if present.

+

Author: Sean Bright + Date: 2023-08-02

+

Fixes #221

+

UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by + the underlying PJSIP library. The bundled version of PJSIP supports + TLS v1.3.

+

func_cut: Add example to documentation.

+

Author: phoneben + Date: 2023-07-19

+

This adds an example to the XML documentation clarifying usage + of the CUT function to address a common misusage.

+

extensions.conf.sample: Remove reference to missing context.

+

Author: Sean Bright + Date: 2023-07-16

+

c3ff4648 removed the [iaxtel700] context but neglected to remove + references to it.

+

This commit addresses that and also removes iaxtel and freeworlddialup + references from other config files.

+

func_export: Use correct function argument as variable name.

+

Author: Sean Bright + Date: 2023-07-12

+

Fixes #208

+

app_queue: Add support for applying caller priority change immediately.

+

Author: Joshua C. Colp + Date: 2023-07-07

+

The app_queue module provides both an AMI action and a CLI command + to change the priority of a caller in a queue. Up to now this change + of priority has only been reflected to new callers into the queue.

+

This change adds an "immediate" option to both the AMI action and + CLI command which immediately applies the priority change respective + to the other callers already in the queue. This can allow, for example, + a caller to be placed at the head of the queue immediately if their + priority is sufficient.

+

Resolves: #202

+

UserNote: The 'queue priority caller' CLI command and + 'QueueChangePriorityCaller' AMI action now have an 'immediate' + argument which allows the caller priority change to be reflected + immediately, causing the position of a caller to move within the + queue depending on the priorities of the other callers.

+

app.h: Move declaration of ast_getdata_result before its first use

+

Author: George Joseph + Date: 2023-07-10

+

The ast_app_getdata() and ast_app_getdata_terminator() declarations + in app.h were changed recently to return enum ast_getdata_result + (which is how they were defined in app.c). The existing + declaration of ast_getdata_result in app.h was about 1000 lines + after those functions however so under certain circumstances, + a "use before declaration" error was thrown by the compiler. + The declaration of the enum was therefore moved to before those + functions.

+

Resolves: #200

+

chan_iax2.c: Avoid crash with IAX2 switch support.

+

Author: Sean Bright + Date: 2023-07-07

+

A change made in 82cebaa0 did not properly handle the case when a + channel was not provided, triggering a crash. ast_check_hangup(...) + does not protect against NULL pointers.

+

Fixes #180

+

res_geolocation: Ensure required 'location_info' is present.

+

Author: Sean Bright + Date: 2023-07-07

+

Fixes #189

+

Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.

+

Author: Mike Bradeen + Date: 2023-06-29

+

Resolves: #181

+

UserNote: The following manager actions have been added

+

VoicemailBoxSummary - Generate message list for a given mailbox

+

VoicemailRemove - Remove a message from a mailbox folder

+

VoicemailMove - Move a message from one folder to another within a mailbox

+

VoicemailForward - Copy a message from one folder in one mailbox + to another folder in another or the same mailbox.

+

app_voicemail: add CLI commands for message manipulation

+

Author: Mike Bradeen + Date: 2023-06-20

+

Adds CLI commands to allow move/remove/forward individual messages + from a particular mailbox folder. The forward command can be used + to copy a message within a mailbox or to another mailbox. Also adds + a show mailbox, required to retrieve message ID's.

+

Resolves: #170

+

UserNote: The following CLI commands have been added to app_voicemail

+

voicemail show mailbox + Show contents of mailbox @

+

voicemail remove + Remove message from in mailbox @

+

voicemail move + Move message in mailbox & from to

+

voicemail forward + Forward message in mailbox @ to + mailbox @

+

res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid into the scope of the rtp_instance lock.

+

Author: zhengsh + Date: 2023-06-30

+

From the gdb information, it was found that when calling __ast_free, the size of the + allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid + is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb, + it is found to be 1.

+

Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid, + which is outside the protection of the rtp_instance lock. However, + ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses + rtp->themssrc_valid within the protection of the rtp_instance lock.

+

This can lead to the possibility that the value of rtp->themssrc_valid used in the call to + ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used + within ast_rtcp_generate_report().

+

Resolves: asterisk#63

+

users.conf: Deprecate users.conf configuration.

+

Author: Naveen Albert + Date: 2023-06-30

+

This deprecates the users.conf config file, which + is no longer as widely supported but still integrated + with a number of different modules.

+

Because there is no real mechanism for marking a + configuration file as "deprecated", and users.conf + is not just used in a single place, this now emits + a warning to the user when the PBX loads to notify + about the deprecation.

+

This configuration mechanism has been widely criticized + and discouraged since its inception, and is no longer + relevant to the configuration that most users are doing + today. Removing it will allow for some simplification + and cleanup in the codebase.

+

Resolves: #183

+

UpgradeNote: The users.conf config is now deprecated + and will be removed in a future version of Asterisk.

+

sig_analog: Allow immediate fake ring to be suppressed.

+

Author: Naveen Albert + Date: 2023-06-08

+

When immediate=yes on an FXS channel, sig_analog will + start fake audible ringback that continues until the + channel is answered. Even if it answers immediately, + the ringback is still audible for a brief moment. + This can be disruptive and unwanted behavior.

+

This adds an option to disable this behavior, though + the default behavior remains unchanged.

+

ASTERISK-30003 #close + Resolves: #118

+

UserNote: The immediatering option can now be set to no to suppress + the fake audible ringback provided when immediate=yes on FXS channels.

+

apply_patches: Use globbing instead of file/sort.

+

Author: Sean Bright + Date: 2023-07-06

+

This accomplishes the same thing as a find ... | sort but with the + added benefit of clarity and avoiding a call to a subshell.

+

Additionally drop the -s option from call to patch as it is not POSIX.

+

apply_patches: Sort patch list before applying

+

Author: George Joseph + Date: 2023-07-06

+

The apply_patches script wasn't sorting the list of patches in + the "patches" directory before applying them. This left the list + in an indeterminate order. In most cases, the list is actually + sorted but rarely, they can be out of order and cause dependent + patches to fail to apply.

+

We now sort the list but the "sort" program wasn't in the + configure scripts so we needed to add that and regenerate + the scripts as well.

+

Resolves: #193

+

pjsip: Upgrade bundled version to pjproject 2.13.1

+

Author: Stanislav Abramenkov + Date: 2023-07-05

+

app_voicemail: fix imap compilation errors

+

Author: Mike Bradeen + Date: 2023-06-26

+

Fixes two compilation errors in app_voicemail_imap, one due to an unsed + variable and one due to a new variable added in the incorrect location + in _163.

+

Resolves: #174

+

res_musiconhold: avoid moh state access on unlocked chan

+

Author: Mike Bradeen + Date: 2023-05-31

+

Move channel unlock to after moh state access to avoid + potential unlocked access to state.

+

Resolves: #133

+

utils: add lock timestamps for DEBUG_THREADS

+

Author: Mike Bradeen + Date: 2023-05-23

+

Adds last locked and unlocked timestamps as well as a + counter for the number of times the lock has been + attempted (vs locked/unlocked) to debug output printed + using the DEBUG_THREADS option.

+

Resolves: #110

+

rest-api: Updates for new documentation site

+

Author: George Joseph + Date: 2023-06-26

+

The new documentation site uses traditional markdown instead + of the Confluence flavored version. This required changes in + the mustache templates and the python that generates the files.

+

rest-api: Ran make ari stubs to fix resource_endpoints inconsistency

+

Author: George Joseph + Date: 2023-06-27

+

app_voicemail_imap: Fix message count when IMAP server is unavailable

+

Author: Olaf Titz + Date: 2023-06-15

+

Some callers of __messagecount did not correctly handle error return, + instead returning a -1 message count. + This caused a notification with "Messages-Waiting: yes" and + "Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable.

+

Fixes: #64

+

res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.

+

Author: Sean Bright + Date: 2023-06-12

+

Resolves: #116

+

Update config.yml

+

Author: Joshua C. Colp + Date: 2023-06-15

+

res_pjsip_session: Added new function calls to avoid ABI issues.

+

Author: Ben Ford + Date: 2023-06-05

+

Added two new functions (ast_sip_session_get_dialog and + ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the + pjsip_inv_state respectively from the pjsip_inv_session on the + ast_sip_session struct. This is due to pjproject adding a new field to + the pjsip_inv_session struct that caused crashes when trying to access + fields that were no longer where they were expected to be if a module + was compiled against a different version of pjproject.

+

Resolves: #145

+

app_queue: Add force_longest_waiting_caller option.

+

Author: Nathan Bruning + Date: 2023-01-24

+

This adds an option 'force_longest_waiting_caller' which changes the + global behavior of the queue engine to prevent queue callers from + 'jumping ahead' when an agent is in multiple queues.

+

Resolves: #108

+

Also closes old asterisk issues: + - ASTERISK-17732 + - ASTERISK-17570

+

pjsip_transport_events.c: Use %zu printf specifier for size_t.

+

Author: Sean Bright + Date: 2023-06-05

+

Partially resolves #143.

+

res_crypto.c: Gracefully handle potential key filename truncation.

+

Author: Sean Bright + Date: 2023-06-05

+

Partially resolves #143.

+

configure: Remove obsolete and deprecated constructs.

+

Author: Sean Bright + Date: 2023-06-01

+

These were uncovered when trying to run bootstrap.sh with Autoconf + 2.71:

+ +

Note that the generated artifacts in this commit are from Autoconf 2.69.

+

Resolves #139

+

res_fax_spandsp.c: Clean up a spaces/tabs issue

+

Author: zhou_jiajian + Date: 2023-05-26

+

ast-db-manage: Synchronize revisions between comments and code.

+

Author: Sean Bright + Date: 2023-06-06

+

In a handful of migrations, the comment header that indicates the + current and previous revisions has drifted from the identifiers + revision and down_revision variables. This updates the comment headers + to match the code.

+

test_statis_endpoints: Fix channel_messages test again

+

Author: George Joseph + Date: 2023-06-12

+

res_crypto.c: Avoid using the non-portable ALLPERMS macro.

+

Author: Sean Bright + Date: 2023-06-05

+

ALLPERMS is not POSIX and it's trivial enough to not jump through + autoconf hoops to check for it.

+

Fixes #149.

+

tcptls: when disabling a server port, we should set the accept_fd to -1.

+

Author: Jaco Kroon + Date: 2023-06-02

+

If we don't set this to -1 if the structure can be potentially re-used + later then it's possible that we'll issue a close() on an unrelated file + descriptor, breaking asterisk in other interesting ways.

+

I believe this to be an unlikely scenario, but it costs nothing to be + safe.

+

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

+

AMI: Add parking position parameter to Park action

+

Author: Jiajian Zhou + Date: 2023-05-19

+

Add a parking space extension parameter (ParkingSpace) to the Park action. + Park action will attempt to park the call to that extension. + If the extension is already in use, then execution will continue at the next priority.

+

UserNote: New ParkingSpace parameter has been added to AMI action Park.

+

test_stasis_endpoints.c: Make channel_messages more stable

+

Author: George Joseph + Date: 2023-06-09

+

The channel_messages test was assuming that stasis would return + messages in a specific order. This is an incorrect assumption as + message ordering was never guaranteed. This was causing the test + to fail occasionally. We now test all the messages for the + required message types instead of testing one by one.

+

Resolves: #158

+

build: Fix a few gcc 13 issues

+

Author: George Joseph + Date: 2023-06-09

+ +

Resolves: #155

+

ast-db-manage: Fix alembic branching error caused by #122.

+

Author: Sean Bright + Date: 2023-06-05

+

Fixes #147.

+

sounds: Update download URL to use HTTPS.

+

Author: Sean Bright + Date: 2023-06-01

+

Related to #136

+

configure: Makefile downloader enable follow redirects.

+

Author: Miguel Angel Nubla + Date: 2023-06-01

+

If curl is used for building, any download such as a sounds package + will fail to follow HTTP redirects and will download wrong data.

+

Resolves: #136

+

res_musiconhold: Add option to loop last file.

+

Author: Naveen Albert + Date: 2023-05-25

+

Adds the loop_last option to res_musiconhold, + which allows the last audio file in the directory + to be looped perpetually once reached, rather than + circling back to the beginning again.

+

Resolves: #122 + ASTERISK-30462

+

UserNote: The loop_last option in musiconhold.conf now + allows the last file in the directory to be looped once reached.

+

chan_dahdi: Fix Caller ID presentation for FXO ports.

+

Author: Naveen Albert + Date: 2023-05-25

+

Currently, the presentation for incoming channels is + always available, because it is never actually set, + meaning the channel presentation can be nonsensical. + If the presentation from the incoming Caller ID spill + is private or unavailable, we now update the channel + presentation to reflect this.

+

Resolves: #120 + ASTERISK-30333 + ASTERISK-21741

+

AMI: Add CoreShowChannelMap action.

+

Author: Ben Ford + Date: 2023-05-18

+

Adds a new AMI action (CoreShowChannelMap) that takes in a channel name + and provides a list of all channels that are connected to that channel, + following local channel connections as well.

+

Resolves: #104

+

UserNote: New AMI action CoreShowChannelMap has been added.

+

sig_analog: Add fuller Caller ID support.

+

Author: Naveen Albert + Date: 2023-05-18

+

A previous change, ASTERISK_29991, made it possible + to send additional Caller ID parameters that were + not previously supported.

+

This change adds support for analog DAHDI channels + to now be able to receive these parameters for + on-hook Caller ID, in order to enhance the usability + of CPE that support these parameters.

+

Resolves: #94 + ASTERISK-30331

+

UserNote: Additional Caller ID properties are now supported on + incoming calls to FXS stations, namely the + redirecting reason and call qualifier.

+

res_stasis.c: Add new type 'sdp_label' for bridge creation.

+

Author: Joe Searle + Date: 2023-05-25

+

Add new type 'sdp_label' when creating a bridge using the ARI. This will + add labels to the SDP for each stream, the label is set to the + corresponding channel id.

+

Resolves: #91

+

UserNote: When creating a bridge using the ARI the 'type' argument now + accepts a new value 'sdp_label' which will configure the bridge to add + labels for each stream in the SDP with the corresponding channel id.

+

app_followme: fix issue with enable_callee_prompt=no (#88)

+

Author: alex2grad + Date: 2023-06-05

+ +

If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk + incorrectly sets a winner channel to the channel from which any control frame was read.

+

This fix sets the winner channel only to the answered channel.

+

Resolves: #87

+

ASTERISK-30326

+

app_queue: Preserve reason for realtime queues

+

Author: Niklas Larsson + Date: 2023-05-05

+

When Asterisk is restarted it does not preserve paused reason for + members of realtime queues. This was fixed for non-realtime queues in + ASTERISK_25732

+

Resolves: #66

+

UpgradeNote: Add a new column to the queue_member table: + reason_paused VARCHAR(80) so the reason can be preserved.

+

UserNote: Make paused reason in realtime queues persist an + Asterisk restart. This was fixed for non-realtime + queues in ASTERISK_25732.

+

indications: logging changes

+

Author: Mike Bradeen + Date: 2023-05-16

+

Increase verbosity to indicate failure due to missing country + and to specify default on CLI dump

+

Resolves: #89

+

callerid: Allow specifying timezone for date/time.

+

Author: Naveen Albert + Date: 2023-05-18

+

The Caller ID generation routine currently is hardcoded + to always use the system time zone. This makes it possible + to optionally specify any TZ-format time zone.

+

Resolves: #98 + ASTERISK-30330

+

logrotate: Fix duplicate log entries.

+

Author: Naveen Albert + Date: 2023-05-18

+

The Asterisk logrotate script contains explicit + references to files with the .log extension, + which are also included when *log is expanded. + This causes issues with newer versions of logrotate. + This fixes this by ensuring that a log file cannot + be referenced multiple times after expansion occurs.

+

Resolves: #96 + ASTERISK-30442 + Reported by: EN Barnett + Tested by: EN Barnett

+

app_sla: Migrate SLA applications out of app_meetme.

+

Author: Naveen Albert + Date: 2023-05-02

+

This removes the dependency of the SLAStation and SLATrunk + applications on app_meetme, in anticipation of the imminent + removal of the deprecated app_meetme module.

+

The user interface for the SLA applications is exactly the + same, and in theory, users should not notice a difference. + However, the SLA applications now use ConfBridge under the + hood, rather than MeetMe, and they are now contained within + their own module.

+

Resolves: #50 + ASTERISK-30309

+

UpgradeNote: The SLAStation and SLATrunk applications have been moved + from app_meetme to app_sla. If you are using these applications and have + autoload=no, you will need to explicitly load this module in modules.conf.

+

chan_pjsip: Allow topology/session refreshes in early media state (#74)

+

Author: Maximilian Fridrich + Date: 2023-05-25

+

With this change, session modifications in the early media state are + possible if the SDP was sent reliably and confirmed by a PRACK. For + details, see RFC 6337, escpecially section 3.2.

+

Resolves: #73

+

chan_dahdi: Fix broken hidecallerid setting. (#101)

+

Author: InterLinked1 + Date: 2023-05-25

+

The hidecallerid setting in chan_dahdi.conf currently + is broken for a couple reasons.

+

First, the actual code in sig_analog to "allow" or "block" + Caller ID depending on this setting improperly used + ast_set_callerid instead of updating the presentation. + This issue was mostly fixed in ASTERISK_29991, and that + fix is carried forward to this code as well.

+

Secondly, the hidecallerid setting is set on the DAHDI + pvt but not carried forward to the analog pvt properly. + This is because the chan_dahdi config loading code improperly + set permhidecallerid to permhidecallerid from the config file, + even though hidecallerid is what is actually set from the config + file. (This is done correctly for call waiting, a few lines above.) + This is fixed to read the proper value.

+

Thirdly, in sig_analog, hidecallerid is set to permhidecallerid + only on hangup. This can lead to potential security vulnerabilities + as an allowed Caller ID from an initial call can "leak" into subsequent + calls if no hangup occurs between them. This is fixed by setting + hidecallerid to permcallerid when calls begin, rather than when they end. + This also means we don't need to also set hidecallerid in chan_dahdi.c + when copying from the config, as we would have to otherwise.

+

Fourthly, sig_analog currently only allows dialing 67 or 82 if + that would actually toggle the presentation. A comment is added + clarifying that this behavior is okay.

+

Finally, a couple log messages are updated to be more accurate.

+

Resolves: #100 + ASTERISK-30349 #close

+

asterisk.c: Fix option warning for remote console. (#103)

+

Author: InterLinked1 + Date: 2023-05-22

+

Commit 09e989f972e2583df4e9bf585c246c37322d8d2f + categorized the T option as not being compatible + with remote consoles, but they do affect verbose + messages with remote console. This fixes this.

+

Resolves: #102

+

res_pjsip_pubsub: Add new pubsub module capabilities. (#82)

+

Author: InterLinked1 + Date: 2023-05-18

+

The existing res_pjsip_pubsub APIs are somewhat limited in + what they can do. This adds a few API extensions that make + it possible for PJSIP pubsub modules to implement richer + features than is currently possible.

+ +

This change does not add any features directly, but makes possible + several new features that will be added in future changes.

+

Resolves: #81 + ASTERISK-30485 #close

+

Master-Only: True

+

configure: fix test code to match gethostbyname_r prototype. (#75)

+

Author: Jaco Kroon + Date: 2023-05-15

+

This enables the test to work with CC=clang.

+

Without this the test for 6 args would fail with:

+

utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration + static int gethostbyname_r (const char name, struct hostent ret, char buf, + ^ + /usr/include/netdb.h:177:12: note: previous declaration is here + extern int gethostbyname_r (const char __restrict __name, + ^

+

Fixing the expected return type to int sorts this out.

+

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

+

res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#47)

+

Author: Sean Bright + Date: 2023-05-11

+

The functionality we are interested in is present only in pjsip 2.13 + and newer.

+

Resolves: #45

+

res_sorcery_memory_cache.c: Fix memory leak (#56)

+

Author: zhengsh + Date: 2023-05-12

+

Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue.

+

Resolves: #55 + ASTERISK-30429

+

utils.h: Deprecate ast_gethostbyname(). (#79)

+

Author: Sean Bright + Date: 2023-05-11

+

Deprecate ast_gethostbyname() in favor of ast_sockaddr_resolve() and + ast_sockaddr_resolve_first_af(). ast_gethostbyname() has not been + used by any in-tree code since 2021.

+

This function will be removed entirely in Asterisk 23.

+

Resolves: #78

+

UpgradeNote: ast_gethostbyname() has been deprecated and will be removed + in Asterisk 23. New code should use ast_sockaddr_resolve() and + ast_sockaddr_resolve_first_af().

+

xml.c: Process XML Inclusions recursively. (#69)

+

Author: Sean Bright + Date: 2023-05-11

+

If processing an XInclude results in new elements, we + need to run XInclude processing again. This continues until no + replacement occurs or an error is encountered.

+

There is a separate issue with dynamic strings (ast_str) that will be + addressed separately.

+

Resolves: #65

+

chan_pjsip: also return all codecs on empty re-INVITE for late offers (#59)

+

Author: Henning Westerholt + Date: 2023-05-04

+

We should also return all codecs on an re-INVITE without SDP for a + call that used late offer (e.g. no SDP in the initial INVITE, SDP + in the ACK). Bugfix for feature introduced in ASTERISK-30193 + (https://issues.asterisk.org/jira/browse/ASTERISK-30193)

+

Migration from previous gerrit change that was not merged.

+

cel: add local optimization begin event (#54)

+

Author: Mike Bradeen + Date: 2023-05-04

+

The current AST_CEL_LOCAL_OPTIMIZE event is and has been + triggered on a local optimization end to serve as a flag + indicating the event occurred. This change adds a second + AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.

+

Resolves: #52

+

UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue + to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event + can be ignored if desired.

+

UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used + by itself or in conert with the existing + AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

+

core: Cleanup gerrit and JIRA references. (#58)

+

Author: Sean Bright + Date: 2023-05-03

+ +

Resolves: #39

+

res_pjsip: mediasec: Add Security-Client headers after 401 (#49)

+

Author: Maximilian Fridrich + Date: 2023-05-02

+

When using mediasec, requests sent after a 401 must still contain the + Security-Client header according to + draft-dawes-sipcore-mediasec-parameter.

+

Resolves: #48

+

LICENSE: Update link to trademark policy. (#44)

+

Author: Joshua C. Colp + Date: 2023-05-02

+

Resolves: #43

+

say.c: Fix French time playback. (#42)

+

Author: InterLinked1 + Date: 2023-05-02

+

ast_waitstream was not called after ast_streamfile, + resulting in "o'clock" being skipped in French.

+

Additionally, the minute announcements should be + feminine.

+

Reported-by: Danny Lloyd

+

Resolves: #41 + ASTERISK-30488

+

chan_dahdi: Add dialmode option for FXS lines.

+

Author: Naveen Albert + Date: 2023-04-28

+

Currently, both pulse and tone dialing are always enabled + on all FXS lines, with no way of disabling one or the other.

+

In some circumstances, it is desirable or necessary to + disable one of these, and this behavior can be problematic.

+

A new "dialmode" option is added which allows setting the + methods to support on a per channel basis for FXS (FXO + signalled lines). The four options are "both", "pulse", + "dtmf"/"tone", and "none".

+

Additionally, integration with the CHANNEL function is + added so that this setting can be updated for a channel + during a call.

+

Resolves: #35 + ASTERISK-29992

+

UserNote: A "dialmode" option has been added which allows + specifying, on a per-channel basis, what methods of + subscriber dialing (pulse and/or tone) are permitted.

+

Additionally, this can be changed on a channel + at any point during a call using the CHANNEL + function.

+

Initial GitHub PRs

+

Author: George Joseph + Date: 2023-04-28

+

Initial GitHub Issue Templates

+

Author: George Joseph + Date: 2023-04-28

+

pbx_dundi: Fix PJSIP endpoint configuration check.

+

Author: Joshua C. Colp + Date: 2023-04-13

+

ASTERISK-28233

+

res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.

+

Author: Naveen Albert + Date: 2023-02-17

+

The current STIR/SHAKEN signing process is inconsistent with the + RFCs in a couple ways that can cause interoperability issues.

+

RFC8225 specifies that the keys must be ordered lexicographically, but + currently the fields are simply ordered according to the order + in which they were added to the JSON object, which is not + compliant with the RFC and can cause issues with some carriers.

+

To fix this, we now leverage libjansson's ability to dump a JSON + object sorted by key value, yielding the correct field ordering.

+

Additionally, telephone numbers must have any leading + prefix removed + and must not contain characters outside of 0-9, *, and # in order + to comply with the RFCs. Numbers are now properly formatted as such.

+

ASTERISK-30407 #close

+

pbx_dundi: Add PJSIP support.

+

Author: Naveen Albert + Date: 2022-12-09

+

Adds PJSIP as a supported technology to DUNDi.

+

To facilitate this, we now allow an endpoint to be specified + for outgoing PJSIP calls. We also allow users to force a specific + channel technology for outgoing SIP-protocol calls.

+

ASTERISK-28109 #close + ASTERISK-28233 #close

+

chan_pjsip: fix music on hold continues after INVITE with replaces

+

Author: Henning Westerholt + Date: 2023-03-21

+

In a three party scenario with INVITE with replaces, we need to + unhold the call, otherwise one party continues to get music on + hold, and the call is not properly bridged between them.

+

ASTERISK-30428

+

install_prereq: Add Linux Mint support.

+

Author: The_Blode + Date: 2023-03-17

+

ASTERISK-30359 #close

+

voicemail.conf: Fix incorrect comment about #include.

+

Author: Naveen Albert + Date: 2023-03-28

+

A comment at the top of voicemail.conf says that #include + cannot be used in voicemail.conf because this breaks + the ability for app_voicemail to auto-update passwords. + This is factually incorrect, since Asterisk has no problem + updating files that are #include'd in the main configuration + file, and this does work in voicemail.conf as well.

+

ASTERISK-30479 #close

+

app_queue: Fix minor xmldoc duplication and vagueness.

+

Author: Naveen Albert + Date: 2023-04-03

+

The F option in the xmldocs for the Queue application + was erroneously duplicated, causing it to display + twice on the wiki. The two sections are now merged into one.

+

Additionally, the description for the d option was quite + vague. Some more details are added to provide context + as to what this actually does.

+

ASTERISK-30486 #close

+

test.c: Fix counting of tests and add 2 new tests

+

Author: George Joseph + Date: 2023-03-28

+

The unit test XML output was counting all registered tests as "run" + even when only a subset were actually requested to be run and + the "failures" attribute was missing.

+ +

Also added 2 new unit tests that just pass and fail to be + used for CI testing.

+

res_pjsip_pubsub: subscription cleanup changes

+

Author: Mike Bradeen + Date: 2023-03-29

+

There are two main parts of the change associated with this + commit. These are driven by the change in call order of + pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject + when an in-dialog SUBSCRIBE is received.

+

First, the previous behavior was for pjproject to call + pubsub_on_rx_refresh before calling pubsub_on_evsub_state + when an in-dialog SUBSCRIBE was received that changes the + subscription state.

+

If that change was a termination due to a re-SUBSCRIBE with + an expires of 0, we used to use the call to pubsub_on_rx_refresh + to set the substate of the evsub to TERMINATE_PENDING before + pjproject could call pubsub_on_evsub_state.

+

This substate let pubsub_on_evsub_state know that the + subscription TERMINATED event could be ignored as there was + still a subsequent NOTIFY that needed to be generated and + another call to pubsub_on_evsub_state to come with it.

+

That NOTIFY was sent via serialized_pubsub_on_refresh_timeout + which would see the TERMINATE_PENDING state and transition it + to TERMINATE_IN_PROGRESS before triggering another call to + pubsub_on_evsub_state (which now would clean up the evsub.)

+

The new pjproject behavior is to call pubsub_on_evsub_state + before pubsub_on_rx_refresh. This means we no longer can set + the state to TERMINATE_PENDING to tell pubsub_on_evsub_state + that it can ignore the first TERMINATED event.

+

To handle this, we now look directly at the event type, + method type and the expires value to determine whether we + want to ignore the event or use it to trigger the evsub + cleanup.

+

Second, pjproject now expects the NOTIFY to actually be sent + during pubsub_on_rx_refresh and avoids the protocol violation + inherent in sending a NOTIFY before the SUBSCRIBE is + acknowledged by caching the sent NOTIFY then sending it + after responding to the SUBSCRIBE.

+

This requires we send the NOTIFY using the non-serialized + pubsub_on_refresh_timeout directly and let pjproject handle + the protocol violation.

+

ASTERISK-30469

+

res_calendar: output busy state as part of show calendar.

+

Author: Jaco Kroon + Date: 2023-03-23

+

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

+

ael: Regenerate lexers and parsers.

+

Author: Sean Bright + Date: 2023-03-21

+

Various changes to ensure that the lexers and parsers can be correctly + generated when REBUILD_PARSERS is enabled.

+

Some notes:

+ +

loader.c: Minor module key check simplification.

+

Author: Sean Bright + Date: 2023-03-23

+

bridge_builtin_features: add beep via touch variable

+

Author: Mike Bradeen + Date: 2023-03-01

+

Add periodic beep option to one-touch recording by setting + the touch variable TOUCH_MONITOR_BEEP or + TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.

+

If the interval is less than 5 seconds, a minimum of 5 + seconds will be imposed. If the interval is set to an + invalid value, it will default to 15 seconds.

+

A new test event PERIODIC_HOOK_ENABLED was added to the + func_periodic_hook hook_on function to indicate when + a hook is started. This is so we can test that the touch + variable starts the hook as expected.

+

ASTERISK-30446

+

res_mixmonitor: MixMonitorMute by MixMonitor ID

+

Author: Mike Bradeen + Date: 2023-03-13

+

While it is possible to create multiple mixmonitor instances + on a channel, it was not previously possible to mute individual + instances.

+

This change includes the ability to specify the MixMonitorID + when calling the manager action: MixMonitorMute. This will + allow an individual MixMonitor instance to be muted via id. + This id can be stored as a channel variable using the 'i' + MixMonitor option.

+

As part of this change, if no MixMonitorID is specified in + the manager action MixMonitorMute, Asterisk will set the mute + flag on all MixMonitor spy-type audiohooks on the channel. + This is done via the new audiohook function: + ast_audiohook_set_mute_all.

+

ASTERISK-30464

+

format_sln: add .slin as supported file extension

+

Author: Mike Bradeen + Date: 2023-03-14

+

Adds '.slin' to existing supported file extensions: + .sln and .raw

+

ASTERISK-30465

+

cli: increase channel column width

+

Author: Mike Bradeen + Date: 2023-03-06

+

For 'core show channels', the Channel name field is increased + to 64 characters and the Location name field is increased to + 32 characters.

+

For 'core show channels verbose', the Channel name field is + increased to 80 characters, the Context is increased to 24 + characters and the Extension is increased to 24 characters.

+

ASTERISK-30455

+

app_osplookup: Remove obsolete sample config.

+

Author: Naveen Albert + Date: 2023-02-24

+

ASTERISK_30302 previously removed app_osplookup, + but its sample config was not removed. + This removes it since nothing else uses it.

+

ASTERISK-30438 #close

+

func_json: Fix JSON parsing issues.

+

Author: Naveen Albert + Date: 2023-02-26

+

Fix issue with returning empty instead of dumping + the JSON string when recursing.

+

Also adds a unit test to capture this fix.

+

ASTERISK-30441 #close

+

app_dial: Fix DTMF not relayed to caller on unanswered calls.

+

Author: Naveen Albert + Date: 2023-03-04

+

DTMF frames are not handled in app_dial when sent towards the + caller. This means that if DTMF is sent to the calling party + and the call has not yet been answered, the DTMF is not audible. + This is now fixed by relaying DTMF frames if only a single + destination is being dialed.

+

ASTERISK-29516 #close

+

configure: fix detection of re-entrant resolver functions

+

Author: Fabrice Fontaine + Date: 2023-03-08

+

uClibc does not provide res_nsearch: + asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'

+

Patch coded by Yann E. MORIN: + http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html

+

ASTERISK-21795 #close

+

Signed-off-by: Bernd Kuhls bernd.kuhls@t-online.de + [Retrieved from: + https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch] + Signed-off-by: Fabrice Fontaine fontaine.fabrice@gmail.com

+

res_agi: RECORD FILE plays 2 beeps.

+

Author: Sean Bright + Date: 2023-03-08

+

Sending the "RECORD FILE" command without the optional + offset_samples argument can result in two beeps playing on the + channel.

+

This bug has been present since Asterisk 0.3.0 (2003-02-06).

+

ASTERISK-30457 #close

+

app_senddtmf: Add SendFlash AMI action.

+

Author: Naveen Albert + Date: 2023-02-26

+

Adds an AMI action to send a flash event + on a channel.

+

ASTERISK-30440 #close

+

http.c: Minor simplification to HTTP status output.

+

Author: Boris P. Korzun + Date: 2023-01-05

+

Change the HTTP status page (located at /httpstatus by default) by:

+ +

ASTERISK-30433 #close

+

make_version: Strip svn stuff and suppress ref HEAD errors

+

Author: George Joseph + Date: 2023-03-13

+ +

res_http_media_cache: Introduce options and customize

+

Author: Holger Hans Peter Freyther + Date: 2022-10-16

+

Make the existing CURL parameters configurable and allow + to specify the usable protocols, proxy and DNS timeout.

+

ASTERISK-30340

+

contrib: rc.archlinux.asterisk uses invalid redirect.

+

Author: Sean Bright + Date: 2023-03-02

+

rc.archlinux.asterisk, which explicitly requests bash in its + shebang, uses the following command syntax:

+
${DAEMON} -rx "core stop now" > /dev/null 2&>1
+
+

The intent of which is to execute:

+
${DAEMON} -rx "core stop now"
+
+

While sending both stdout and stderr to /dev/null. Unfortunately, + because the & is in the wrong place, bash is interpreting the 2 as + just an additional argument to the $DAEMON command and not as a file + descriptor and proceeds to use the bashism &> to send stderr and + stdout to a file named 1.

+

So we clean it up and just use bash's shortcut syntax.

+

Issue raised and a fix suggested (but not used) by peutch on GitHub¹.

+

ASTERISK-30449 #close

+
    +
  1. https://github.com/asterisk/asterisk/pull/31
    2. +
+

main/iostream.c: fix build with libressl

+

Author: Fabrice Fontaine + Date: 2023-02-25

+

Fix the following build failure with libressl by using SSL_is_server + which is available since version 2.7.0 and + https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:

+

iostream.c: In function 'ast_iostream_close': + iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'} + 559 | if (!stream->ssl->server) { + | ^~

+

ASTERISK-30107 #close

+

Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9

+

res_pjsip: Replace invalid UTF-8 sequences in callerid name

+

Author: George Joseph + Date: 2023-02-16

+ +

ASTERISK-27830

+

test.c: Avoid passing -1 to FD_* family of functions.

+

Author: Sean Bright + Date: 2023-02-27

+

This avoids buffer overflow errors when running tests that capture + output from child processes.

+

This also corrects a copypasta in an off-nominal error message.

+

chan_iax2: Fix jitterbuffer regression prior to receiving audio.

+

Author: Naveen Albert + Date: 2022-12-14

+

ASTERISK_29392 (a security fix) introduced a regression by + not processing frames when we don't have an audio format.

+

Currently, chan_iax2 only calls jb_get to read frames from + the jitterbuffer when the voiceformat has been set on the pvt. + However, this only happens when we receive a voice frame, which + means that prior to receiving voice frames, other types of frames + get stalled completely in the jitterbuffer.

+

To fix this, we now fallback to using the format negotiated during + call setup until we've actually received a voice frame with a format. + This ensures we're always able to read from the jitterbuffer.

+

ASTERISK-30354 #close + ASTERISK-30162 #close

+

test_crypto.c: Fix getcwd(…) build error.

+

Author: Sean Bright + Date: 2023-02-27

+

getcwd(…) is decorated with the warn_unused_result attribute and + therefore needs its return value checked.

+

pjproject_bundled: fix cross-compilation with ssl libs

+

Author: Nick French + Date: 2023-02-11

+

Asterisk makefiles auto-detect ssl library availability, + then they assume that pjproject makefiles will also autodetect + an ssl library at the same time, so they do not pass on the + autodetection result to pjproject.

+

This normally works, except the pjproject makefiles disables + autodetection when cross-compiling.

+

Fix by explicitly configuring pjproject to use ssl if we + have been told to use it or it was autodetected

+

ASTERISK-30424 #close

+

res_phoneprov.c: Multihomed SERVER cache prevention

+

Author: cmaj + Date: 2023-01-07

+

Phones moving between subnets on multi-homed server have their + initially connected interface IP cached in the SERVER variable, + even when it is not specified in the configuration files. This + prevents phones from obtaining the correct SERVER variable value + when they move to another subnet.

+

ASTERISK-30388 #close + Reported-by: cmaj

+

app_read: Add an option to return terminator on empty digits.

+

Author: Mike Bradeen + Date: 2023-01-30

+

Adds 'e' option to allow Read() to return the terminator as the + dialed digits in the case where only the terminator is entered.

+

ie; if "#" is entered, return "#" if the 'e' option is set and "" + if it is not.

+

ASTERISK-30411

+

app_directory: Add a 'skip call' option.

+

Author: Mike Bradeen + Date: 2023-01-27

+

Adds 's' option to skip calling the extension and instead set the + extension as DIRECTORY_EXTEN channel variable.

+

ASTERISK-30405

+

app_senddtmf: Add option to answer target channel.

+

Author: Mike Bradeen + Date: 2023-02-06

+

Adds a new option to SendDTMF() which will answer the specified + channel if it is not already up. If no channel is specified, the + current channel will be answered instead.

+

ASTERISK-30422

+

res_pjsip: Prevent SEGV in pjsip_evsub_send_request

+

Author: Mike Bradeen + Date: 2023-02-21

+

contributed pjproject - patch to check sub->pending_notify + in evsub.c:on_tsx_state before calling + pjsip_evsub_send_request()

+

res_pjsip_pubsub - change post pjsip 2.13 behavior to use + pubsub_on_refresh_timeout to avoid the ao2_cleanup call on + the sub_tree. This is is because the final NOTIFY send is no + longer the last place the sub_tree is referenced.

+

ASTERISK-30419

+

app_queue: Minor docs and logging fixes for UnpauseQueueMember.

+

Author: Sean Bright + Date: 2023-02-02

+

ASTERISK-30417 #close

+

app_queue: Reset all queue defaults before reload.

+

Author: Sean Bright + Date: 2023-01-31

+

Several queue fields were not being set to their default value during + a reload.

+

Additionally added some sample configuration options that were missing + from queues.conf.sample.

+

res_pjsip: Upgraded bundled pjsip to 2.13

+

Author: Mike Bradeen + Date: 2023-01-20

+

Removed multiple patches.

+

Code chages in res_pjsip_pubsub due to changes in evsub.

+

Pjsip now calls on_evsub_state() before on_rx_refresh(), + so the sub tree deletion that used to take place in + on_evsub_state() now must take place in on_rx_refresh().

+

Additionally, pjsip now requires that you send the NOTIFY + from within on_rx_refresh(), otherwise it will assert + when going to send the 200 OK. The idea is that it will + look for this NOTIFY and cache it until after sending the + response in order to deal with the self-imposed message + mis-order. Asterisk previously dealt with this by pushing + the NOTIFY in on_rx_refresh(), but pjsip now forces us + to use it's method.

+

Changes were required to configure in order to detect + which way pjsip handles this as the two are not + compatible for the reasons mentioned above.

+

A corresponding change in testsuite is required in order + to deal with the small interal timing changes caused by + moving the NOTIFY send.

+

ASTERISK-30325

+

doxygen: Fix doxygen errors.

+

Author: Sean Bright + Date: 2023-01-30

+

app_signal: Add signaling applications

+

Author: Naveen Albert + Date: 2022-01-06

+

Adds the Signal and WaitForSignal + applications, which can be used for inter-channel + signaling in the dialplan.

+

Signal supports sending a signal to other channels + listening for a signal of the same name, with an + optional data payload. The signal is received by + all channels waiting for that named signal.

+

ASTERISK-29810 #close

+

app_directory: add ability to specify configuration file

+

Author: Mike Bradeen + Date: 2023-01-25

+

Adds option to app_directory to specify a filename from which to + read configuration instead of voicemail.conf ie;

+

same => n,Directory(,,c(directory.conf))

+

This configuration should contain a list of extensions using the + voicemail.conf format, ie;

+

2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no

+

ASTERISK-30404

+

func_json: Enhance parsing capabilities of JSON_DECODE

+

Author: Naveen Albert + Date: 2022-02-12

+

Adds support for arrays to JSON_DECODE by allowing the + user to print out entire arrays or index a particular + key or print the number of keys in a JSON array.

+

Additionally, adds support for recursively iterating a + JSON tree in a single function call, making it easier + to parse JSON results with multiple levels. A maximum + depth is imposed to prevent potentially blowing + the stack.

+

Also fixes a bug with the unit tests causing an empty + string to be printed instead of the actual test result.

+

ASTERISK-29913 #close

+

res_pjsip_session: Add overlap_context option.

+

Author: Naveen Albert + Date: 2022-10-13

+

Adds the overlap_context option, which can be used + to explicitly specify a context to use for overlap + dialing extension matches, rather than forcibly + using the context configured for the endpoint.

+

ASTERISK-30262 #close

+

res_stasis_snoop: Fix snoop crash

+

Author: sungtae kim + Date: 2023-01-04

+

Added NULL pointer check and channel lock to prevent resource release + while the chanspy is processing.

+

ASTERISK-29604

+

res_monitor: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-18

+

ASTERISK-30303

+

app_playback.c: Fix PLAYBACKSTATUS regression.

+

Author: Sean Bright + Date: 2023-01-05

+

In Asterisk 11, if a channel was redirected away during Playback(), + the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 + (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that + behavior was inadvertently changed and the same operation would result + in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 + behavior has been restored.

+

Partial fix for ASTERISK~25661.

+

res_rtp_asterisk: Don't use double math to generate timestamps

+

Author: George Joseph + Date: 2023-01-11

+

Rounding issues with double math were causing rtp timestamp + slips in outgoing packets. We're now back to integer math + and are getting no more slips.

+

ASTERISK-30391

+

app_macro: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-12-12

+

For most modules that interacted with app_macro, this change is limited + to no longer looking for the current context from the macrocontext when + set. Additionally, the following modules are impacted:

+

app_dial - no longer supports M^ connected/redirecting macro + app_minivm - samples written using macro will no longer work. + The sample needs a re-write

+

app_queue - can no longer a macro on the called party's channel. + Use gosub which is currently supported

+

ccss - no callback macro, gosub only

+

app_voicemail - no macro support

+

channel - remove macrocontext and priority, no connected line or + redirection macro options + options - stdexten is deprecated to gosub as the default and only + pbx - removed macrolock + pbx_dundi - no longer look for macro

+

snmp - removed macro context, exten, and priority

+

ASTERISK-30304

+

format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)

+

Author: Alexei Gradinari + Date: 2023-01-06

+

Each playback of WAV files results in logging + "Skipping unknown block 'LIST'".

+

To prevent unnecessary flooding of this DEBUG log this patch replaces + ast_log(LOG_DEBUG, ...) by ast_debug(1, ...).

+

res_pjsip_rfc3326: Add SIP causes support for RFC3326

+

Author: Igor Goncharovsky + Date: 2022-11-18

+

Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).

+

ASTERISK-30319 #close

+

res_rtp_asterisk: Asterisk Media Experience Score (MES)

+

Author: George Joseph + Date: 2022-10-28

+
+

This commit reinstates MES with some casting fixes to the + functions in time.h that convert between doubles and timeval + structures. The casting issues were causing incorrect + timestamps to be calculated which caused transcoding from/to + G722 to produce bad or no audio.

+

ASTERISK-30391

+
+

This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is avilable using + the same mechanisms you'd use to retrieve jitter, loss, + and rtt statistics. For more information about the + score and how to retrieve it, see + https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

+ +

Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"

+

Author: George Joseph + Date: 2023-01-09

+

This reverts commit e66c5da145b4545428fca768db7fb0921156af98.

+

Reason for revert: Issue when transcoding to/from g722

+

http.c: Fix NULL pointer dereference bug

+

Author: Boris P. Korzun + Date: 2022-12-28

+

If native HTTP is disabled but HTTPS is enabled and status page enabled + too, Core/HTTP crashes while loading. 'global_http_server' references + to NULL, but the status page tries to dereference it.

+

The patch adds a check for HTTP is enabled.

+

ASTERISK-30379 #close

+

loader: Allow declined modules to be unloaded.

+

Author: Naveen Albert + Date: 2022-12-08

+

Currently, if a module declines to load, dlopen is called + to register the module but dlclose never gets called. + Furthermore, loader.c currently doesn't allow dlclose + to ever get called on the module, since it declined to + load and the unload function bails early in this case.

+

This can be problematic if a module is updated, since the + new module cannot be loaded into memory since we haven't + closed all references to it. To fix this, we now allow + modules to be unloaded, even if they never "loaded" in + Asterisk itself, so that dlclose is called and the module + can be properly cleaned up, allowing the updated module + to be loaded from scratch next time.

+

ASTERISK-30345 #close

+

app_broadcast: Add Broadcast application

+

Author: Naveen Albert + Date: 2022-08-15

+

Adds a new application, Broadcast, which can be used for + one-to-many transmission and many-to-one reception of + channel audio in Asterisk. This is similar to ChanSpy, + except it is designed for multiple channel targets instead + of a single one. This can make certain kinds of audio + manipulation more efficient and streamlined. New kinds + of audio injection impossible with ChanSpy are also made + possible.

+

ASTERISK-30180 #close

+

func_frame_trace: Print text for text frames.

+

Author: Naveen Albert + Date: 2022-12-13

+

Since text frames contain a text body, make FRAME_TRACE + more useful for text frames by actually printing the text.

+

ASTERISK-30353 #close

+

app_cdr: Remove deprecated application and option.

+

Author: Naveen Albert + Date: 2022-12-22

+

This removes the deprecated NoCDR application, which + was deprecated in Asterisk 12, having long been fully + superseded by the CDR_PROP function.

+

The deprecated e option to ResetCDR is also removed + for the same reason.

+

ASTERISK-30371 #close

+

res_http_media_cache: Do not crash when there is no extension

+

Author: Holger Hans Peter Freyther + Date: 2022-12-16

+

Do not crash when a URL has no path component as in this case the + ast_uri_path function will return NULL. Make the code cope with not + having a path.

+

The below would crash

+
+

media cache create http://google.com /tmp/foo.wav

+
+

Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault. + 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 + (gdb) bt + #0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 + #1 0x0000ffff43d43a78 in file_extension_from_string (str=, buffer=buffer@entry=0xffffca9973c0 "", + capacity=capacity@entry=64) at res_http_media_cache.c:288 + #2 0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568, + buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378 + #3 0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392 + #4 0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555 + #5 0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=, data=, object=) + at res_http_media_cache.c:613 + #6 0x0000000000487638 in bucket_file_wizard_create (sorcery=, data=, object=) + at bucket.c:191 + #7 0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718, + details=details@entry=0xffffca9974a8) at sorcery.c:2027 + #8 0x0000000000559698 in ast_sorcery_create (sorcery=, object=object@entry=0x3bf96568) at sorcery.c:2077 + #9 0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727 + #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com", + file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335 + #11 0x00000000004f88ec in media_cache_handle_create_item (e=, cmd=, a=0xffffca9976b8) + at media_cache.c:640

+

ASTERISK-30375 #close

+

manager: Fix appending variables.

+

Author: Naveen Albert + Date: 2022-12-22

+

The if statement here is always false after the for + loop finishes, so variables are never appended. + This removes that to properly append to the end + of the variable list.

+

ASTERISK-30351 #close + Reported by: Sebastian Gutierrez

+

json.h: Add ast_json_object_real_get.

+

Author: Naveen Albert + Date: 2022-12-16

+

json.h contains macros to get a string and an integer + from a JSON object. However, the macro to do this for + JSON reals is missing. This adds that.

+

ASTERISK-30361 #close

+

res_pjsip_transport_websocket: Add remote port to transport

+

Author: George Joseph + Date: 2022-12-23

+

When Asterisk receives a new websocket conenction, it creates a new + pjsip transport for it and copies connection data into it. The + transport manager then uses the remote IP address and port on the + transport to create a monitor for each connection. However, the + remote port wasn't being copied, only the IP address which meant + that the transport manager was creating only 1 monitoring entry for + all websocket connections from the same IP address. Therefore, if + one of those connections failed, it deleted the transport taking + all the the connections from that same IP address with it.

+ +

ASTERISK-30369

+

chan_sip: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-28

+

ASTERISK-30297

+

res_rtp_asterisk: Asterisk Media Experience Score (MES)

+

Author: George Joseph + Date: 2022-10-28

+

This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is avilable using + the same mechanisms you'd use to retrieve jitter, loss, + and rtt statistics. For more information about the + score and how to retrieve it, see + https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

+ +

ASTERISK-30280

+

pbx_app: Update outdated pbx_exec channel snapshots.

+

Author: Naveen Albert + Date: 2022-12-21

+

pbx_exec makes a channel snapshot before executing applications. + This doesn't cause an issue during normal dialplan execution + where pbx_exec is called over and over again in succession. + However, if pbx_exec is called "one off", e.g. using + ast_pbx_exec_application, then a channel snapshot never ends + up getting made after the executed application returns, and + inaccurate snapshot information will linger for a while, causing + "core show channels", etc. to show erroneous info.

+

This is fixed by manually making a channel snapshot at the end + of ast_pbx_exec_application, since we anticipate that pbx_exec + might not get called again immediately.

+

ASTERISK-30367 #close

+

res_pjsip_session: Use Caller ID for extension matching.

+

Author: Naveen Albert + Date: 2022-11-26

+

Currently, there is no Caller ID available to us when + checking for an extension match when handling INVITEs. + As a result, extension patterns that depend on the Caller ID + are not matched and calls may be incorrectly rejected.

+

The Caller ID is not available because the supplement that + adds Caller ID to the session does not execute until after + this check. Supplement callbacks cannot yet be executed + at this point since the session is not yet in the appropriate + state.

+

To fix this without impacting existing behavior, the Caller ID + number is now retrieved before attempting to pattern match. + This ensures pattern matching works correctly and there is + no behavior change to the way supplements are called.

+

ASTERISK-28767 #close

+

pbx_builtins: Remove deprecated and defunct functionality.

+

Author: Naveen Albert + Date: 2022-11-29

+

This removes the ImportVar and SetAMAFlags applications + which have been deprecated since Asterisk 12, but were + never removed previously.

+

Additionally, it removes remnants of defunct options + that themselves were removed years ago.

+

ASTERISK-30335 #close

+

res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.

+

Author: Ben Ford + Date: 2022-12-12

+

When a call is put on hold and it has moh_passthrough and rtp_timeout + set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is + expected to be used, but rtp_timeout is used instead. This change adds a + couple of checks for locally_held to determine if rtp_timeout_hold needs + to be used instead of rtp_timeout.

+

ASTERISK-30350

+

app_voicemail_odbc: Fix string overflow warning.

+

Author: Naveen Albert + Date: 2022-11-14

+

Fixes a negative offset warning by initializing + the buffer to empty.

+

Additionally, although it doesn't currently complain + about it, the size of a buffer is increased to + accomodate the maximum size contents it could have.

+

ASTERISK-30240 #close

+

streams: Ensure that stream is closed in ast_stream_and_wait on error

+

Author: Peter Fern + Date: 2022-11-22

+

When ast_stream_and_wait returns an error (for example, when attempting + to stream to a channel after hangup) the stream is not closed, and + callers typically do not check the return code. This results in leaking + file descriptors, leading to resource exhaustion.

+

This change ensures that the stream is closed in case of error.

+

ASTERISK-30198 #close + Reported-by: Julien Alie

+

func_callerid: Warn about invalid redirecting reason.

+

Author: Naveen Albert + Date: 2022-11-26

+

Currently, if a user attempts to set a Caller ID related + function to an invalid value, a warning is emitted, + except for when setting the redirecting reason. + We now emit a warning if we were unable to successfully + parse the user-provided reason.

+

ASTERISK-30332 #close

+

app_sendtext: Remove references to removed applications.

+

Author: Naveen Albert + Date: 2022-12-10

+

Removes see-also references to applications that don't + exist anymore (removed in Asterisk 19), + so these dead links don't show up on the wiki.

+

ASTERISK-30347 #close

+

res_pjsip: Fix path usage in case dialing with '@'

+

Author: Igor Goncharovsky + Date: 2022-11-04

+

Fix aor lookup on sip path addition. Issue happens in case of dialing + with @ and overriding user part of RURI.

+

ASTERISK-30100 #close + Reported-by: Yury Kirsanov

+

res_geoloc: fix NULL pointer dereference bug

+

Author: Alexandre Fournier + Date: 2022-12-09

+

The ast_geoloc_datastore_add_eprofile function does not return 0 on + success, it returns the size of the underlying datastore. This means + that the datastore will be freed and its pointer set to NULL when no + error occured at all.

+

ASTERISK-30346

+

res_pjsip_aoc: Don't assume a body exists on responses.

+

Author: Joshua C. Colp + Date: 2022-12-13

+

When adding AOC to an outgoing response the code + assumed that a body would exist for comparing the + Content-Type. This isn't always true.

+

The code now checks to make sure the response has + a body before checking the Content-Type.

+

ASTERISK-21502

+

app_if: Fix format truncation errors.

+

Author: Naveen Albert + Date: 2022-12-12

+

Fixes format truncation warnings in gcc 12.2.1.

+

ASTERISK-30349 #close

+

chan_alsa: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-14

+

ASTERISK-30298

+

manager: AOC-S support for AOCMessage

+

Author: Michael Kuron + Date: 2022-11-01

+

ASTERISK-21502

+

chan_mgcp: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-15

+

Also removes res_pktcops to avoid merge conflicts + with ASTERISK~30301.

+

ASTERISK-30299

+

res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip

+

Author: Michael Kuron + Date: 2022-10-23

+

chan_sip supported sending AOC-D and AOC-E information in SIP INFO + messages in an "AOC" header in a format that was originally defined by + Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC + format that is supported by devices from multiple vendors, including + Snom phones with firmware >= 8.4.2 (released in 2010).

+

This commit adds a new res_pjsip_aoc module that inserts AOC information + into outgoing messages or sends SIP INFO messages as described below. + It also fixes a small issue in res_pjsip_session which didn't always + call session supplements on outgoing_response.

+ +

The specification defines one more, AOC-S in an INVITE request, which + is not implemented here because it is not currently possible in + Asterisk to have AOC data ready at this point in call setup. Once + specifying AOC-S via the dialplan or passing it through from another + SIP channel's INVITE is possible, that might be added.

+

The SIP INFO requests are sent out immediately when the AOC indication + is received. The others are inserted into an appropriate outgoing + message whenever that is ready to be sent. In the latter case, the XML + is stored in a channel variable at the time the AOC indication is + received. Depending on where the AOC indications are coming from (e.g. + PRI or AMI), it may not always be possible to guarantee that the AOC-E + is available in time for the BYE.

+

Successfully tested AOC-D and both variants of AOC-E with a Snom D735 + running firmware 10.1.127.10. It does not appear to properly support + AOC-S however, so that could only be tested by inspecting SIP traces.

+

ASTERISK-21502 #close + Reported-by: Matt Jordan mjordan@digium.com

+

res_hep: Add support for named capture agents.

+

Author: Naveen Albert + Date: 2022-11-21

+

Adds support for the capture agent name field + of the Homer protocol to Asterisk by allowing + users to specify a name that will be sent to + the HEP server.

+

ASTERISK-30322 #close

+

res_pjsip: Fix typo in from_domain documentation

+

Author: Marcel Wagner + Date: 2022-11-25

+

This fixes a small typo in the from_domain documentation on the endpoint documentation

+

ASTERISK-30328 #close

+

app_if: Adds conditional branch applications

+

Author: Naveen Albert + Date: 2021-06-28

+

Adds the If, ElseIf, Else, ExitIf, and EndIf + applications for conditional execution + of a block of dialplan, similar to the While, + EndWhile, and ExitWhile applications. The + appropriate branch is executed at most once + if available and may be broken out of while + inside.

+

ASTERISK-29497

+

res_pjsip_session.c: Map empty extensions in INVITEs to s.

+

Author: Naveen Albert + Date: 2022-10-17

+

Some SIP devices use an empty extension for PLAR functionality.

+

Rather than rejecting these empty extensions, we now use the s + extension for such calls to mirror the existing PLAR functionality + in Asterisk (e.g. chan_dahdi).

+

ASTERISK-30265 #close

+

res_pjsip: Update contact_user to point out default

+

Author: Marcel Wagner + Date: 2022-11-17

+

Updates the documentation for the 'contact_user' field to point out the + default outbound contact if no contact_user is specified 's'

+

ASTERISK-30316 #close

+

res_pjsip_header_funcs: Add custom parameter support.

+

Author: Naveen Albert + Date: 2022-07-21

+

Adds support for custom URI and header parameters + in the From header in PJSIP. Parameters can be + both set and read using this function.

+

ASTERISK-30150 #close

+

app_voicemail: Fix missing email in msg_create_from_file.

+

Author: Naveen Albert + Date: 2022-11-03

+

msg_create_from_file currently does not dispatch emails, + which means that applications using this function, such + as MixMonitor, will not trigger notifications to users + (only AMI events are sent our currently). This is inconsistent + with other ways users can receive voicemail.

+

This is fixed by adding an option that attempts to send + an email and falling back to just the notifications as + done now if that fails. The existing behavior remains + the default.

+

ASTERISK-30283 #close

+

ari: Destroy body variables in channel create.

+

Author: Joshua C. Colp + Date: 2022-12-08

+

When passing a JSON body to the 'create' channel route + it would be converted into Asterisk variables, but never + freed resulting in a memory leak.

+

This change makes it so that the variables are freed in + all cases.

+

ASTERISK-30344

+

res_adsi: Fix major regression caused by media format rearchitecture.

+

Author: Naveen Albert + Date: 2022-11-23

+

The commit that rearchitected media formats, + a2c912e9972c91973ea66902d217746133f96026 (ASTERISK_23114) + introduced a regression by improperly translating code in res_adsi.c. + In particular, the pointer to the frame buffer was initialized + at the top of adsi_careful_send, rather than dynamically updating it + for each frame, as is required.

+

This resulted in the first frame being repeatedly sent, + rather than advancing through the frames. + This corrupted the transmission of the CAS to the CPE, + which meant that CPE would never respond with the DTMF acknowledgment, + effectively completely breaking ADSI functionality.

+

This issue is now fixed, and ADSI now works properly again.

+

ASTERISK-29793 #close

+

func_presencestate: Fix invalid memory access.

+

Author: Naveen Albert + Date: 2022-11-13

+

When parsing information from AstDB while loading, + it is possible that certain pointers are never + set, which leads to invalid memory access and + then, fatally, invalid free attempts on this memory. + We now initialize to NULL to prevent this.

+

ASTERISK-30311 #close

+

sig_analog: Fix no timeout duration.

+

Author: Naveen Albert + Date: 2022-12-01

+

ASTERISK_28702 previously attempted to fix an + issue with flash hook hold timing out after + just under 17 minutes, when it should have never + been timing out. It fixed this by changing 999999 + to INT_MAX, but it did so in chan_dahdi, which + is the wrong place since ss_thread is now in + sig_analog and the one in chan_dahdi is mostly + dead code.

+

This fixes this by porting the fix to sig_analog.

+

ASTERISK-30336 #close

+

xmldoc: Allow XML docs to be reloaded.

+

Author: Naveen Albert + Date: 2022-11-05

+

The XML docs are currently only loaded on + startup with no way to update them during runtime. + This makes it impossible to load modules that + use ACO/Sorcery (which require documentation) + if they are added to the source tree and built while + Asterisk is running (e.g. external modules).

+

This adds a CLI command to reload the XML docs + during runtime so that documentation can be updated + without a full restart of Asterisk.

+

ASTERISK-30289 #close

+

rtp_engine.h: Update examples using ast_format_set.

+

Author: Naveen Albert + Date: 2022-11-24

+

This file includes some doxygen comments referencing + ast_format_set. This is an obsolete API that was + removed years back, but documentation was not fully + updated to reflect that. These examples are + updated to the current way of doing things + (using the format cache).

+

ASTERISK-30327 #close

+

app_osplookup: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-18

+

ASTERISK-30302

+

chan_skinny: Remove deprecated module.

+

Author: Mike Bradeen + Date: 2022-11-16

+

ASTERISK-30300

+

app_mixmonitor: Add option to use real Caller ID for voicemail.

+

Author: Naveen Albert + Date: 2022-11-04

+

MixMonitor currently uses the Connected Line as the Caller ID + for voicemails. This is due to the implementation being written + this way for use with Digium phones. However, in general this + is not correct for generic usage in the dialplan, and people + may need the real Caller ID instead. This adds an option to do that.

+

ASTERISK-30286 #close

+

manager: prevent file access outside of config dir

+

Author: Mike Bradeen + Date: 2022-10-03

+

Add live_dangerously flag to manager and use this flag to + determine if a configuation file outside of AST_CONFIG_DIR + should be read.

+

ASTERISK-30176

+

pjsip_transport_events: Fix possible use after free on transport

+

Author: George Joseph + Date: 2022-10-10

+

It was possible for a module that registered for transport monitor + events to pass in a pjsip_transport that had already been freed. + This caused pjsip_transport_events to crash when looking up the + monitor for the transport. The fix is a two pronged approach.

+
    +
  1. +

    We now increment the reference count on pjsip_transports when we + create monitors for them, then decrement the count when the + transport is going to be destroyed.

    +
    2. +
  2. +

    There are now APIs to register and unregister monitor callbacks + by "transport key" which is a string concatenation of the remote ip + address and port. This way the module needing to monitor the + transport doesn't have to hold on to the transport object itself to + unregister. It just has to save the transport_key.

    +
    3. +
  3. +

    Added the pjsip_transport reference increment and decrement.

    +
    4. +
  4. +

    Changed the internal transport monitor container key from the + transport->obj_name (which may not be unique anyway) to the + transport_key.

    +
    5. +
  5. +

    Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that + fills a buffer with the transport_key using a passed-in + pjsip_transport.

    +
    6. +
  6. +

    Added the following functions: + ast_sip_transport_monitor_register_key + ast_sip_transport_monitor_register_replace_key + ast_sip_transport_monitor_unregister_key + and marked their non-key counterparts as deprecated.

    +
    7. +
  7. +

    Updated res_pjsip_pubsub and res_pjsip_outbound_register to use + the new "key" monitor functions.

    +
    8. +
+

NOTE: res_pjsip_registrar also uses the transport monitor + functionality but doesn't have a persistent object other than + contact to store a transport key. At this time, it continues to + use the non-key monitor functions.

+

ASTERISK-30244

+

pjproject: 2.13 security fixes

+

Author: Ben Ford + Date: 2022-11-29

+

Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13.

+

ASTERISK-30338

+

pbx_builtins: Allow Answer to return immediately.

+

Author: Naveen Albert + Date: 2022-11-11

+

The Answer application currently waits for up to 500ms + for media, even if users specify a different timeout.

+

This adds an option to not wait for media on the channel + by doing a raw answer instead. The default 500ms threshold + is also documented.

+

ASTERISK-30308 #close

+

chan_dahdi: Allow FXO channels to start immediately.

+

Author: Naveen Albert + Date: 2022-11-11

+

Currently, chan_dahdi will wait for at least one + ring before an incoming call can enter the dialplan. + This is generally necessary in order to receive + the Caller ID spill and/or distinctive ringing + detection.

+

However, if neither of these is required, then there + is nothing gained by waiting for one ring and this + unnecessarily delays call setup. Users can now + use immediate=yes to make FXO channels (FXS signaled) + begin processing dialplan as soon as Asterisk receives + the call.

+

ASTERISK-30305 #close

+

sla: Prevent deadlock and crash due to autoservicing.

+

Author: Naveen Albert + Date: 2022-09-24

+

SLAStation currently autoservices the station channel before + creating a thread to actually dial the trunk. This leads + to duplicate servicing of the channel which causes assertions, + deadlocks, crashes, and moreover not the correct behavior.

+

Removing the autoservice prevents the crash, but if the station + hangs up before the trunk answers, the call hangs since the hangup + was never serviced on the channel.

+

This is fixed by not autoservicing the channel, but instead + servicing it in the thread dialing the trunk, since it is doing + so synchronously to begin with. Instead of sleeping for 100ms + in a loop, we simply use the channel for timing, and abort + if it disappears.

+

The same issue also occurs with SLATrunk when a call is answered, + because ast_answer invokes ast_waitfor_nandfds. Thus, we use + ast_raw_answer instead which does not cause any conflict and allows + the call to be answered normally without thread blocking issues.

+

ASTERISK-29998 #close

+

Build system: Avoid executable stack.

+

Author: Jaco Kroon + Date: 2022-11-07

+

Found in res_geolocation, but I believe others may have similar issues, + thus not linking to a specific issue.

+

Essentially gcc doesn't mark the stack for being non-executable unless + it's compiling the source, this informs ld via gcc to mark the object as + not requiring an executable stack (which a binary blob obviously + doesn't).

+

ASTERISK-30321

+

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

+

func_json: Fix memory leak.

+

Author: Naveen Albert + Date: 2022-11-10

+

A memory leak was present in func_json due to + using ast_json_free, which just calls ast_free, + as opposed to recursively freeing the JSON + object as needed. This is now fixed to use the + right free functions.

+

ASTERISK-30293 #close

+

test_json: Remove duplicated static function.

+

Author: Naveen Albert + Date: 2022-11-10

+

Removes the function mkstemp_file and uses + ast_file_mkftemp from file.h instead.

+

ASTERISK-30295 #close

+

res_agi: Respect "transmit_silence" option for "RECORD FILE".

+

Author: Joshua C. Colp + Date: 2022-11-16

+

The "RECORD FILE" command in res_agi has its own + implementation for actually doing the recording. This + has resulted in it not actually obeying the option + "transmit_silence" when recording.

+

This change causes it to now send silence if the + option is enabled.

+

ASTERISK-30314

+

file.c: Don't emit warnings on winks.

+

Author: Naveen Albert + Date: 2022-11-06

+

Adds an ignore case for wink since it should + pass through with no warning.

+

ASTERISK-30290 #close

+

app_mixmonitor: Add option to delete files on exit.

+

Author: Naveen Albert + Date: 2022-11-03

+

Adds an option that allows MixMonitor to delete + its copy of any recording files before exiting.

+

This can be handy in conjunction with options + like m, which copy the file elsewhere, and the + original files may no longer be needed.

+

ASTERISK-30284 #close

+

translate.c: Prefer better codecs upon translate ties.

+

Author: Naveen Albert + Date: 2021-05-27

+

If multiple codecs are available for the same + resource and the translation costs between + multiple codecs are the same, ties are + currently broken arbitrarily, which means a + lower quality codec would be used. This forces + Asterisk to explicitly use the higher quality + codec, ceteris paribus.

+

ASTERISK-29455

+

manager: Update ModuleCheck documentation.

+

Author: Naveen Albert + Date: 2022-11-03

+

The ModuleCheck XML documentation falsely + claims that the module's version number is returned. + This has not been the case since 14, since the version + number is not available anymore, but the documentation + was not changed at the time. It is now updated to + reflect this.

+

ASTERISK-30285 #close

+

runUnittests.sh: Save coredumps to proper directory

+

Author: George Joseph + Date: 2022-11-02

+

Fixed the specification of "outputdir" when calling ast_coredumper + so the txt files are saved in the correct place.

+

ASTERISK-30282

+

chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer

+

Author: George Joseph + Date: 2022-11-02

+

unicast_rtp_request() was setting the channel variables like this:

+

pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", + ast_sockaddr_stringify_addr(&local_address)); + ast_rtp_instance_get_local_address(instance, &local_address); + pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", + ast_sockaddr_stringify_port(&local_address));

+

...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being + set before local_address was set. In fact, the address part of + local_address was set earlier in the function, just not the port. + This was confusing however so ast_rtp_instance_get_local_address() + is now being called before setting UNICASTRTP_LOCAL_ADDRESS.

+

ASTERISK-30281

+

res_pjsip: prevent crash on websocket disconnect

+

Author: Mike Bradeen + Date: 2022-10-13

+

When a websocket (or potentially any stateful connection) is quickly + created then destroyed, it is possible that the qualify thread will + destroy the transaction before the initialzing thread is finished + with it.

+

Depending on the timing, this can cause an assertion within pjsip.

+

To prevent this, ast_send_stateful_response will now create the group + lock and add a reference to it before creating the transaction.

+

While this should resolve the crash, there is still the potential that + the contact will not be cleaned up properly, see:ASTERISK~29286. As a + result, the contact has to 'time out' before it will be removed.

+

ASTERISK-28689

+

tcptls: Prevent crash when freeing OpenSSL errors.

+

Author: Naveen Albert + Date: 2022-10-27

+

write_openssl_error_to_log has been erroneously + using ast_free instead of free, which will + cause a crash when MALLOC_DEBUG is enabled since + the memory was not allocated by Asterisk's memory + manager. This changes it to use the actual free + function directly to avoid this.

+

ASTERISK-30278 #close

+

res_pjsip_outbound_registration: Allow to use multiple proxies for registration

+

Author: Igor Goncharovsky + Date: 2022-09-09

+

Current registration code use pjsip_parse_uri to verify outbound_proxy + that is different from the reading this option for the endpoint. This + made value with multiple proxies invalid for registration pjsip settings. + Removing URI validation helps to use registration through multiple proxies.

+

ASTERISK-30217 #close

+

tests: Fix compilation errors on 32-bit.

+

Author: Naveen Albert + Date: 2022-10-23

+

Fix compilation errors caused by using size_t + instead of uintmax_t and non-portable format + specifiers.

+

ASTERISK-30273 #close

+

res_pjsip: return all codecs on a re-INVITE without SDP

+

Author: Henning Westerholt + Date: 2022-08-26

+

Currently chan_pjsip on receiving a re-INVITE without SDP will only + return the codecs that are previously negotiated and not offering + all enabled codecs.

+

This causes interoperability issues with different equipment (e.g. + from Cisco) for some of our customers and probably also in other + scenarios involving 3PCC infrastructure.

+

According to RFC 3261, section 14.2 we SHOULD return all codecs + on a re-INVITE without SDP

+

The PR proposes a new parameter to configure this behaviour: + all_codecs_on_empty_reinvite. It includes the code, documentation, + alembic migrations, CHANGES file and example configuration additions.

+

ASTERISK-30193 #close

+

res_pjsip_notify: Add option support for AMI.

+

Author: Naveen Albert + Date: 2022-10-14

+

The PJSIP notify CLI commands allow for using + "options" configured in pjsip_notify.conf.

+

This allows these same options to be used in + AMI actions as well.

+

Additionally, as part of this improvement, + some repetitive common code is refactored.

+

ASTERISK-30263 #close

+

res_pjsip_logger: Add method-based logging option.

+

Author: Naveen Albert + Date: 2022-07-21

+

Expands the pjsip logger to support the ability to filter + by SIP message method. This can make certain types of SIP debugging + easier by only logging messages of particular method(s).

+

ASTERISK-30146 #close

+

Co-authored-by: Sean Bright sean@seanbright.com

+

Dialing API: Cancel a running async thread, may not cancel all calls

+

Author: Frederic LE FOLL + Date: 2022-10-06

+

race condition: ast_dial_join() may not cancel outgoing call, if + function is called just after called party answer and before + application execution (bit is_running_app not yet set).

+

This fix adds ast_softhangup() calls in addition to existing + pthread_kill() when is_running_app is not set.

+

ASTERISK-30258

+

chan_dahdi: Fix unavailable channels returning busy.

+

Author: Naveen Albert + Date: 2022-10-23

+

This fixes dahdi_request to properly set the cause + code to CONGESTION instead of BUSY if no channels + were actually available.

+

Currently, the cause is erroneously set to busy + if the channel itself is found, regardless of its + current state. However, if the channel is not available + (e.g. T1 down, card not operable, etc.), then the + channel itself may not be in a functional state, + in which case CHANUNAVAIL is the correct cause to use.

+

This adds a simple check to ensure that busy tone + is only returned if a channel is encountered that + has an owner, since that is the only possible way + that a channel could actually be busy.

+

ASTERISK-30274 #close

+

res_pjsip_pubsub: Prevent removing subscriptions.

+

Author: Naveen Albert + Date: 2022-10-16

+

pjproject does not provide any mechanism of removing + event packages, which means that once a subscription + handler is registered, it is effectively permanent.

+

pjproject will assert if the same event package is + ever registered again, so currently unloading and + loading any Asterisk modules that use subscriptions + will cause a crash that is beyond our control.

+

For that reason, we now prevent users from being + able to unload these modules, to prevent them + from ever being loaded twice.

+

ASTERISK-30264 #close

+

say: Don't prepend ampersand erroneously.

+

Author: Naveen Albert + Date: 2022-09-28

+

Some logic in say.c for determining if we need + to also add an ampersand for file seperation was faulty, + as non-successful files would increment the count, causing + a leading ampersand to be added improperly.

+

This is fixed, and a unit test that captures this regression + is also added.

+

ASTERISK-30248 #close

+

res_crypto: handle unsafe private key files

+

Author: Philip Prindeville + Date: 2022-09-16

+

ASTERISK-30213 #close

+

audiohook: add directional awareness

+

Author: Mike Bradeen + Date: 2022-09-29

+

Add enum to allow setting optional direction. If set to only one + direction, only feed matching-direction frames to the associated + slin factory.

+

This prevents mangling the transcoder on non-mixed frames when the + READ and WRITE frames would have otherwise required it. Also + removes the need to mute or discard the un-wanted frames as they + are no longer added in the first place.

+

res_stasis_snoop is changed to use this addition to set direction + on audiohook based on spy direction.

+

If no direction is set, the ast_audiohook_init will init this enum + to BOTH which maintains existing functionality.

+

ASTERISK-30252

+

cdr: Allow bridging and dial state changes to be ignored.

+

Author: Naveen Albert + Date: 2022-06-01

+

Allows bridging, parking, and dial messages to be globally + ignored for all CDRs such that only a single CDR record + is generated per channel.

+

This is useful when CDRs should endure for the lifetime of + an entire channel and bridging and dial updates in the + dialplan should not result in multiple CDR records being + created for the call. With the ignore bridging option, + bridging changes have no impact on the channel's CDRs. + With the ignore dial state option, multiple Dials and their + outcomes have no impact on the channel's CDRs. The + last disposition on the channel is preserved in the CDR, + so the actual disposition of the call remains available.

+

These two options can reduce the amount of "CDR hacks" that + have hitherto been necessary to ensure that CDR was not + "spoiled" by these messages if that was undesired, such as + putting a dummy optimization-disabled local channel between + the caller and the actual call and putting the CDR on the channel + in the middle to ensure that CDR would persist for the entire + call and properly record start, answer, and end times. + Enabling these options is desirable when calls correspond + to the entire lifetime of channels and the CDR should + reflect that.

+

Current default behavior remains unchanged.

+

ASTERISK-30091 #close

+

res_tonedetect: Add ringback support to TONE_DETECT.

+

Author: Naveen Albert + Date: 2022-09-30

+

Adds support for detecting audible ringback tone + to the TONE_DETECT function using the p option.

+

ASTERISK-30254 #close

+

chan_dahdi: Resolve format truncation warning.

+

Author: Naveen Albert + Date: 2022-10-01

+

Fixes a format truncation warning in notify_message.

+

ASTERISK-30256 #close

+

res_crypto: don't modify fname in try_load_key()

+

Author: Philip Prindeville + Date: 2022-09-16

+

"fname" is passed in as a const char , but strstr() mangles that + into a char , and we were attempting to modify the string in place. + This is an unwanted (and undocumented) side-effect.

+

ASTERISK-30213

+

res_crypto: use ast_file_read_dirs() to iterate

+

Author: Philip Prindeville + Date: 2022-09-15

+

ASTERISK-30213

+

res_geolocation: Update wiki documentation

+

Author: George Joseph + Date: 2022-09-27

+

Also added a note to the geolocation.conf.sample file + and added a README to the res/res_geolocation/wiki + directory.

+

res_pjsip: Add mediasec capabilities.

+

Author: Maximilian Fridrich + Date: 2022-07-26

+

This patch adds support for mediasec SIP headers and SDP attributes. + These are defined in RFC 3329, 3GPP TS 24.229 and + draft-dawes-sipcore-mediasec-parameter. The new features are + implemented so that a backbone for RFC 3329 is present to streamline + future work on RFC 3329.

+

With this patch, Asterisk can communicate with Deutsche Telekom trunks + which require these fields.

+

ASTERISK-30032

+

res_prometheus: Do not crash on invisible bridges

+

Author: Holger Hans Peter Freyther + Date: 2022-09-20

+

Avoid crashing by skipping invisible bridges and checking the + snapshot for a null pointer. In effect this is how the bridges + are enumerated in res/ari/resource_bridges.c already.

+

ASTERISK-30239 + ASTERISK-30237

+

db: Fix incorrect DB tree count for AMI.

+

Author: Naveen Albert + Date: 2022-09-24

+

The DBGetTree AMI action's ListItem previously + always reported 1, regardless of the count. This + is corrected to report the actual count.

+

ASTERISK-30245 #close + patches: + gettreecount.diff submitted by Birger Harzenetter (license 5870)

+

res_pjsip_geolocation: Change some notices to debugs.

+

Author: Naveen Albert + Date: 2022-09-19

+

If geolocation is not in use for an endpoint, the NOTICE + log level is currently spammed with messages about this, + even though nothing is wrong and these messages provide + no real value. These log messages are therefore changed + to debugs.

+

ASTERISK-30241 #close

+

func_logic: Don't emit warning if both IF branches are empty.

+

Author: Naveen Albert + Date: 2022-09-21

+

The IF function currently emits warnings if both IF branches + are empty. However, there is no actual necessity that either + branch be non-empty as, unlike other conditional applications/ + functions, nothing is inherently done with IF, and both + sides could legitimately be empty. The warning is thus turned + into a debug message.

+

ASTERISK-30243 #close

+

features: Add no answer option to Bridge.

+

Author: Naveen Albert + Date: 2022-09-11

+

Adds the n "no answer" option to the Bridge application + so that answer supervision can not automatically + be provided when Bridge is executed.

+

Additionally, a mechanism (dialplan variable) + is added to prevent bridge targets (typically the + target of a masquerade) from answering the channel + when they enter the bridge.

+

ASTERISK-30223 #close

+

app_bridgewait: Add option to not answer channel.

+

Author: Naveen Albert + Date: 2022-09-09

+

Adds the n option to not answer the channel when calling + BridgeWait, so the application can be used without + forcing answer supervision.

+

ASTERISK-30216 #close

+

app_amd: Add option to play audio during AMD.

+

Author: Naveen Albert + Date: 2022-08-15

+

Adds an option that will play an audio file + to the party while AMD is running on the + channel, so the called party does not just + hear silence.

+

ASTERISK-30179 #close

+

test: initialize capture structure before freeing

+

Author: Philip Prindeville + Date: 2022-09-15

+

ASTERISK-30232 #close

+

func_export: Add EXPORT function

+

Author: Naveen Albert + Date: 2021-05-17

+

Adds the EXPORT function, which allows write + access to variables and functions on other + channels.

+

ASTERISK-29432 #close

+

res_pjsip: Add 100rel option "peer_supported".

+

Author: Maximilian Fridrich + Date: 2022-07-26

+

This patch adds a new option to the 100rel parameter for pjsip + endpoints called "peer_supported". When an endpoint with this option + receives an incoming request and the request indicated support for the + 100rel extension, then Asterisk will send 1xx responses reliably. If + the request did not indicate 100rel support, Asterisk sends 1xx + responses normally.

+

ASTERISK-30158

+

manager: be more aggressive about purging http sessions.

+

Author: Jaco Kroon + Date: 2022-09-05

+

If we find that n_max (currently hard wired to 1) sessions were purged, + schedule the next purge for 1ms into the future rather than 5000ms (as + per current). This way we will purge up to 1000 sessions per second + rather than 1 every 5 seconds.

+

This mitigates a build-up of sessions should http sessions gets + established faster than 1 per 5 seconds.

+

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

+

func_scramble: Fix null pointer dereference.

+

Author: Naveen Albert + Date: 2022-09-10

+

Fix segfault due to null pointer dereference + inside the audiohook callback.

+

ASTERISK-30220 #close

+

func_strings: Add trim functions.

+

Author: Naveen Albert + Date: 2022-09-11

+

Adds TRIM, LTRIM, and RTRIM, which can be used + for trimming leading and trailing whitespace + from strings.

+

ASTERISK-30222 #close

+

res_crypto: Memory issues and uninitialized variable errors

+

Author: George Joseph + Date: 2022-09-16

+

ASTERISK-30235

+

res_geolocation: Fix issues exposed by compiling with -O2

+

Author: George Joseph + Date: 2022-09-16

+

Fixed "may be used uninitialized" errors in geoloc_config.c.

+

ASTERISK-30234

+

res_crypto: don't complain about directories

+

Author: Philip Prindeville + Date: 2022-09-13

+

ASTERISK-30226 #close

+

res_pjsip: Add user=phone on From and PAID for usereqphone=yes

+

Author: Mike Bradeen + Date: 2022-08-15

+

Adding user=phone to local-side uri's when user_eq_phone=yes is set for + an endpoint. Previously this would only add the header to the To and R-URI.

+

ASTERISK-30178

+

res_geolocation: Fix segfault when there's an empty element

+

Author: George Joseph + Date: 2022-09-13

+

Fixed a segfault caused by var_list_from_loc_info() encountering + an empty location info element.

+

Fixed an issue in ast_strsep() where a value with only whitespace + wasn't being preserved.

+

Fixed an issue in ast_variable_list_from_quoted_string() where + an empty value was considered a failure.

+

ASTERISK-30215 + Reported by: Dan Cropp

+

res_musiconhold: Add option to not play music on hold on unanswered channels

+

Author: sungtae kim + Date: 2022-08-14

+

This change adds an option, answeredonly, that will prevent music on + hold on channels that are not answered.

+

ASTERISK-30135

+

res_pjsip: Add TEL URI support for basic calls.

+

Author: Ben Ford + Date: 2022-08-02

+

This change allows TEL URI requests to come through for basic calls. The + allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To + headers will now allow TEL URIs, as well as the request URI.

+

Support is only for TEL URIs present in traffic from a remote party. + Asterisk does not generate any TEL URIs on its own.

+

ASTERISK-26894

+

res_crypto: Use EVP API's instead of legacy API's

+

Author: Philip Prindeville + Date: 2022-03-24

+

ASTERISK-30046 #close

+

test: Add coverage for res_crypto

+

Author: Philip Prindeville + Date: 2022-05-03

+

We're validating the following functionality:

+

encrypting a block of data with RSA + decrypting a block of data with RSA + signing a block of data with RSA + verifying a signature with RSA + encrypting a block of data with AES-ECB + encrypting a block of data with AES-ECB

+

as well as accessing test keys from the keystore.

+

ASTERISK-30045 #close

+

res_crypto: make keys reloadable on demand for testing

+

Author: Philip Prindeville + Date: 2022-07-26

+

ASTERISK-30045

+

test: Add test coverage for capture child process output

+

Author: Philip Prindeville + Date: 2022-05-03

+

ASTERISK-30037 #close

+

main/utils: allow checking for command in $PATH

+

Author: Philip Prindeville + Date: 2022-07-26

+

ASTERISK-30037

+

test: Add ability to capture child process output

+

Author: Philip Prindeville + Date: 2022-05-02

+

ASTERISK-30037

+

res_crypto: Don't load non-regular files in keys directory

+

Author: Philip Prindeville + Date: 2022-04-26

+

ASTERISK-30046

+

func_frame_trace: Remove bogus assertion.

+

Author: Naveen Albert + Date: 2022-09-08

+

The FRAME_TRACE function currently asserts if it sees + a MASQUERADE_NOTIFY. However, this is a legitimate thing + that can happen so asserting is inappropriate, as there + are no clear negative ramifications of such a thing. This + is adjusted to be like the other frames to print out + the subclass.

+

ASTERISK-30210 #close

+

lock.c: Add AMI event for deadlocks.

+

Author: Naveen Albert + Date: 2022-07-27

+

Adds an AMI event to indicate that a deadlock + has likely started, when Asterisk is compiled + with DETECT_DEADLOCKS enabled. This can make + it easier to perform automated deadlock detection + and take appropriate action (such as doing a core + dump). Unlike the deadlock warnings, the AMI event + is emitted only once per deadlock.

+

ASTERISK-30161 #close

+

app_confbridge: Add end_marked_any option.

+

Author: Naveen Albert + Date: 2022-09-04

+

Adds the end_marked_any option, which can be used + to kick a user from a conference if any marked user + leaves.

+

ASTERISK-30211 #close

+

pbx_variables: Use const char if possible.

+

Author: Naveen Albert + Date: 2022-09-03

+

Use const char for char arguments to + pbx_substitute_variables_helper_full_location + that can do so (context and exten).

+

ASTERISK-30209 #close

+

res_geolocation: Add two new options to GEOLOC_PROFILE

+

Author: George Joseph + Date: 2022-08-25

+

Added an 'a' option to the GEOLOC_PROFILE function to allow + variable lists like location_info_refinement to be appended + to instead of replacing the entire list.

+

Added an 'r' option to the GEOLOC_PROFILE function to resolve all + variables before a read operation and after a Set operation.

+

Added a few missing parameters to the ones allowed for writing + with GEOLOC_PROFILE.

+

Fixed a bug where calling GEOLOC_PROFILE to read a parameter + might actually update the profile object.

+

Cleaned up XML documentation a bit.

+

ASTERISK-30190

+

res_geolocation: Allow location parameters on the profile object

+

Author: George Joseph + Date: 2022-08-18

+

You can now specify the location object's format, location_info, + method, location_source and confidence parameters directly on + a profile object for simple scenarios where the location + information isn't common with any other profiles. This is + mutually exclusive with setting location_reference on the + profile.

+

Updated appdocsxml.dtd to allow xi:include in a configObject + element. This makes it easier to link to complete configOptions + in another object. This is used to add the above fields to the + profile object without having to maintain the option descriptions + in two places.

+

ASTERISK-30185

+

res_geolocation: Add profile parameter suppress_empty_ca_elements

+

Author: George Joseph + Date: 2022-08-17

+

Added profile parameter "suppress_empty_ca_elements" that + will cause Civic Address elements that are empty to be + suppressed from the outgoing PIDF-LO document.

+

Fixed a possible SEGV if a sub-parameter value didn't have a + value.

+

ASTERISK-30177

+

res_geolocation: Add built-in profiles

+

Author: George Joseph + Date: 2022-08-16

+

The trigger to perform outgoing geolocation processing is the + presence of a geoloc_outgoing_call_profile on an endpoint. This + is intentional so as to not leak location information to + destinations that shouldn't receive it. In a totally dynamic + configuration scenario however, there may not be any profiles + defined in geolocation.conf. This makes it impossible to do + outgoing processing without defining a "dummy" profile in the + config file.

+

This commit adds 4 built-in profiles: + "" + "" + "" + "" + The profiles are empty except for having their precedence + set and can be set on an endpoint to allow processing without + entries in geolocation.conf. "" is actually the + best one to use in this situation.

+

ASTERISK-30182

+

res_pjsip_sdp_rtp: Skip formats without SDP details.

+

Author: Joshua C. Colp + Date: 2022-08-30

+

When producing an outgoing SDP we iterate through the configured + formats and produce SDP information. It is possible for some + configured formats to not have SDP information available. If this + is the case we skip over them to allow the SDP to still be + produced.

+

ASTERISK-29185

+

cli: Prevent assertions on startup from bad ao2 refs.

+

Author: Naveen Albert + Date: 2022-05-03

+

If "core show channels" is run before startup has completed, it + is possible for bad ao2 refs to occur because the system is not + yet fully initialized. This will lead to an assertion failing.

+

To prevent this, initialization of CLI builtins is moved to be + later along in the main load sequence. Core CLI commands are + loaded at the same time, but channel-related commands are loaded + later on.

+

ASTERISK-29846 #close

+

pjsip: Add TLS transport reload support for certificate and key.

+

Author: Joshua C. Colp + Date: 2022-08-19

+

This change adds support using the pjsip_tls_transport_restart + function for reloading the TLS certificate and key, if the filenames + remain unchanged. This is useful for Let's Encrypt and other + situations. Note that no restart of the transport will occur if + the certificate and key remain unchanged.

+

ASTERISK-30186

+

res_tonedetect: Fix typos referring to wrong variables.

+

Author: Naveen Albert + Date: 2022-08-25

+

Fixes two typos that cause fax detection to not work. + One refers to the wrong frame variable, and the other + refers to the subclass.integer instead of the frametype + as it should.

+

ASTERISK-30192 #close

+

alembic: add missing ps_endpoints columns

+

Author: Mike Bradeen + Date: 2022-08-17

+

The following required columns were missing, + now added to the ps_endpoints table:

+

incoming_call_offer_pref + outgoing_call_offer_pref + stir_shaken_profile

+

ASTERISK-29453

+

chan_dahdi.c: Resolve a format-truncation build warning.

+

Author: Sean Bright + Date: 2022-08-19

+

With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0:

+
+

chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated + writing up to 255 bytes into a region of size between 242 and 252 + [-Werror=format-truncation=]

+
+

This removes the error-prone sizeof(...) calculations in favor of just + doubling the size of the base buffer.

+

res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update

+

Author: Alexei Gradinari + Date: 2022-08-03

+

Set termination state to old subscriptions to prevent queueing and sending + NOTIFY messages on exten/device state changes.

+

Postpone destruction of old subscriptions until all already queued tasks + that may be using old subscriptions have completed.

+

ASTERISK-29906

+

channel.h: Remove redundant declaration.

+

Author: Sean Bright + Date: 2022-08-15

+

The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set() + for us, so no need to declare it separately.

+

features: Add transfer initiation options.

+

Author: Naveen Albert + Date: 2022-02-05

+

Adds additional control options over the transfer + feature functionality to give users more control + in how the transfer feature sounds and works.

+

First, the "transfer" sound that plays when a transfer is + initiated can now be customized by the user in + features.conf, just as with the other transfer sounds.

+

Secondly, the user can now specify the transfer extension + in advance by using the TRANSFER_EXTEN variable. If + a valid extension is contained in this variable, the call + will automatically be transferred to this destination. + Otherwise, it will fall back to collecting the extension + from the user as is always done now.

+

ASTERISK-29899 #close

+

CI: Fixing path issue on venv check

+

Author: Mike Bradeen + Date: 2022-08-31

+

ASTERISK-26826

+

CI: use Python3 virtual environment

+

Author: Mike Bradeen + Date: 2022-08-11

+

Requires Python3 testsuite changes

+

ASTERISK-26826

+

general: Very minor coding guideline fixes.

+

Author: Naveen Albert + Date: 2022-07-28

+

Fixes a few coding guideline violations: + * Use of C99 comments + * Opening brace on same line as function prototype

+

ASTERISK-30163 #close

+

res_geolocation: Address user issues, remove complexity, plug leaks

+

Author: George Joseph + Date: 2022-08-05

+ +

ASTERISK-30167

+

chan_iax2: Add missing options documentation.

+

Author: Naveen Albert + Date: 2022-07-30

+

Adds missing dial resource option documentation.

+

ASTERISK-30164 #close

+

app_confbridge: Fix memory leak on updated menu options.

+

Author: Naveen Albert + Date: 2022-08-01

+

If the CONFBRIDGE function is used to dynamically set + menu options, a memory leak occurs when a menu option + that has been set is overridden, since the menu entry + is not destroyed before being freed. This ensures that + it is.

+

Additionally, logic that duplicates the destroy function + is removed in lieu of the destroy function itself.

+

ASTERISK-28422 #close

+

Geolocation: Wiki Documentation

+

Author: George Joseph + Date: 2022-07-19

+

manager: Remove documentation for nonexistent action.

+

Author: Naveen Albert + Date: 2022-07-28

+

The manager XML documentation documents a "FilterList" + action, but there is no such action. Therefore, this can + lead to confusion when people try to use a documented + action that does not, in fact, exist. This is removed + as the action never did exist in the past, nor would it + be trivial to add since we only store the regex_t + objects, so the filter list can't actually be provided + without storing that separately. Most likely, the + documentation was originally added (around version 10) + in anticipation of something that never happened.

+

ASTERISK-29917 #close

+

cdr.conf: Remove obsolete app_mysql reference.

+

Author: Naveen Albert + Date: 2022-07-27

+

The CDR sample config still mentions that app_mysql + is available in the addons directory, but this is + incorrect as it was removed as of 19. This removes + that to avoid confusion.

+

ASTERISK-30160 #close

+

general: Remove obsolete SVN references.

+

Author: Naveen Albert + Date: 2022-07-27

+

There are a handful of files in the tree that + reference an SVN link for the coding guidelines.

+

This removes these because the links are dead + and the vast majority of source files do not + contain these links, so this is more consistent.

+

app_skel still maintains an (up to date) link + to the coding guidelines.

+

ASTERISK-30159 #close

+

app_meetme: Add missing AMI documentation.

+

Author: Naveen Albert + Date: 2022-07-23

+

The MeetmeList and MeetmeListRooms AMI + responses are currently completely undocumented. + This adds documentation for these responses.

+

ASTERISK-30018 #close

+

general: Improve logging levels of some log messages.

+

Author: Naveen Albert + Date: 2022-07-22

+

Adjusts some logging levels to be more or less important, + that is more prominent when actual problems occur and less + prominent for less noteworthy things.

+

ASTERISK-30153 #close

+

app_confbridge: Add missing AMI documentation.

+

Author: Naveen Albert + Date: 2022-07-23

+

Documents the ConfbridgeListRooms AMI response, + which is currently not documented.

+

ASTERISK-30020 #close

+

func_srv: Document field parameter.

+

Author: Naveen Albert + Date: 2022-07-23

+

Adds missing documentation for the field parameter + for the SRVRESULT function.

+

ASTERISK-30151 + Reported by: Chris Young

+

pbx_functions.c: Manually update ast_str strlen.

+

Author: Naveen Albert + Date: 2022-07-23

+

When ast_func_read2 is used to read a function using + its read function (as opposed to a native ast_str read2 + function), the result is copied directly by the function + into the ast_str buffer. As a result, the ast_str length + remains initialized to 0, which is a bug because this is + not the real string length.

+

This can cascade and have issues elsewhere, such as when + reading substrings of functions that only register read + as opposed to read2 callbacks. In this case, since reading + ast_str_strlen returns 0, the returned substring is empty + as opposed to the actual substring. This has caused + the ast_str family of functions to behave inconsistently + and erroneously, in contrast to the pbx_variables substitution + functions which work correctly.

+

This fixes this issue by manually updating the ast_str length + when the result is copied directly into the ast_str buffer.

+

Additionally, an assertion and a unit test that previously + exposed these issues are added, now that the issue is fixed.

+

ASTERISK-29966 #close

+

build: fix bininstall launchd issue on cross-platform build

+

Author: Sergey V. Lobanov + Date: 2022-02-19

+

configure script detects /sbin/launchd, but the result of this + check is not used in Makefile (bininstall). Makefile also detects + /sbin/launchd file to decide if it is required to install + safe_asterisk.

+

configure script correctly detects cross compile build and sets + PBX_LAUNCHD=0

+

In case of building asterisk on MacOS host for Linux target using + external toolchain (e.g. OpenWrt toolchain), bininstall does not + install safe_asterisk (due to /sbin/launchd detection in Makefile), + but it is required on target (Linux).

+

This patch adds HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@ to makeopts.in to + use the result of /sbin/launchd detection from configure script in + Makefile. + Also this patch uses HAVE_SBIN_LAUNCHD in Makefile (bininstall) to + decide if it is required to install safe_asterisk.

+

ASTERISK-29905 #close

+

manager: Fix incomplete filtering of AMI events.

+

Author: Naveen Albert + Date: 2022-07-12

+

The global event filtering code was only in one + possible execution path, so not all events were + being properly filtered out if requested. This moves + that into the universal AMI handling code so all + events are properly handled.

+

Additionally, the CLI listing of disabled events can + also get truncated, so we now print out everything.

+

ASTERISK-30137 #close

+

db: Add AMI action to retrieve DB keys at prefix.

+

Author: Naveen Albert + Date: 2022-07-11

+

Adds the DBGetTree action, which can be used to + retrieve all of the DB keys beginning with a + particular prefix, similar to the capability + provided by the database show CLI command.

+

ASTERISK-30136 #close

+

Update master branch for Asterisk 21

+

Author: George Joseph + Date: 2022-07-20

+ diff --git a/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.md b/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.md new file mode 100644 index 0000000000..8d42f36a6c --- /dev/null +++ b/ChangeLogs/ChangeLog-certified-22.8-cert1-rc1.md @@ -0,0 +1,15527 @@ + +## Change Log for Release asterisk-certified-22.8-cert1-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-22.8-cert1-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/certified-20.7-cert9...certified-22.8-cert1-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-22.8-cert1-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/certified-asterisk) + +### Summary: + +- Commits: 853 +- Commit Authors: 110 +- Issues Resolved: 590 +- Security Advisories Resolved: 13 + - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw): Using malformed From header can forge identity with ";" or NULL in name portion + - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh): Path traversal via AMI ListCategories allows access to outside files + - [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j): A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash + - [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42): Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection + - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44): Write=originate, is sufficient permissions for code execution / System() dialplan + - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2): cli_permissions.conf: deny option does not work for disallowing shell commands + - [GHSA-hxj9-xwr8-w8pq](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq): Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation + - [GHSA-mrq5-74j5-f5cr](https://github.com/asterisk/asterisk/security/advisories/GHSA-mrq5-74j5-f5cr): Remote DoS and possible RCE in asterisk/res/res_stir_shaken/verification.c + - [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3): ast_coredumper running as root sources ast_debug_tools.conf from /etc/asterisk; potentially leading to privilege escalation + - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9): A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used + - [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh): The Asterisk embedded web server's /httpstatus page echos user supplied values(cookie and query string) without sanitization + - [GHSA-v9q8-9j8m-5xwp](https://github.com/asterisk/asterisk/security/advisories/GHSA-v9q8-9j8m-5xwp): Uncontrolled Search-Path Element in safe_asterisk script may allow local privilege escalation. + - [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c): ast_coredumper runs as root, and writes gdb init file to world writeable folder; leading to potential privilege escalation + +### User Notes: + +- #### ast_coredumper: check ast_debug_tools.conf permissions + ast_debug_tools.conf must be owned by root and not be + writable by other users or groups to be used by ast_coredumper or + by ast_logescalator or ast_loggrabber when run as root. + +- #### chan_websocket.conf.sample: Fix category name. + The category name in the chan_websocket.conf.sample file was + incorrect. It should be "global" instead of "general". + +- #### cli.c: Allow 'channel request hangup' to accept patterns. + The 'channel request hangup' CLI command now accepts + multiple channel names, POSIX Extended Regular Expressions, glob-like + patterns, or a combination of all of them. See the CLI command 'core + show help channel request hangup' for full details. + +- #### res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache populate command + The AMI command sorcery memory cache populate will now + return an error if there is an internal error performing the populate. + The CLI command will display an error in this case as well. + +- #### res_geolocation: Fix multiple issues with XML generation. + Geolocation: Two new optional profile parameters have been added. + * `pidf_element_id` which sets the value of the `id` attribute on the top-level + PIDF-LO `device`, `person` or `tuple` elements. + * `device_id` which sets the content of the `` element. + Both parameters can include channel variables. + +- #### res_pjsip_messaging: Add support for following 3xx redirects + A new pjsip endpoint option follow_redirect_methods was added. + This option is a comma-delimited, case-insensitive list of SIP methods + for which SIP 3XX redirect responses are followed. An alembic upgrade + script has been added for adding this new option to the Asterisk + database. + +- #### taskprocessors: Improve logging and add new cli options + New CLI command has been added - + core show taskprocessor name + +- #### ccss: Add option to ccss.conf to globally disable it. + A new "enabled" parameter has been added to ccss.conf. It defaults + to "yes" to preserve backwards compatibility but CCSS is rarely used so + setting "enabled = no" in the "general" section can save some unneeded channel + locking operations and log message spam. Disabling ccss will also prevent + the func_callcompletion and chan_dahdi modules from loading. + +- #### Makefile: Add module-list-* targets. + Try "make module-list-deprecated" to see what modules + are on their way out the door. + +- #### app_mixmonitor: Add 's' (skip) option to delay recording. + This change introduces a new 's()' (skip) option to the MixMonitor + application. Example: + MixMonitor(${UNIQUEID}.wav,s(3)) + This skips recording for the first 3 seconds before writing audio to the file. + Existing MixMonitor behavior remains unchanged when the 's' option is not used. + +- #### app_queue.c: Only announce to head caller if announce_to_first_user + When announce_to_first_user is false, no announcements are played to the head caller + +- #### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. + The STIR_SHAKEN_ATTESTATION dialplan function has been added + which will allow suppressing attestation on a call-by-call basis + regardless of the profile attached to the outgoing endpoint. + +- #### func_channel: Allow R/W of ADSI CPE capability setting. + CHANNEL(adsicpe) can now be read or written to change + the channels' ADSI CPE capability setting. + +- #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() + Added a new option to HANGUPCAUSE to access additional + information about hangup reason. Reason headers from pjsip + could be read using 'tech_extended' cause type. + +- #### func_math: Add DIGIT_SUM function. + The DIGIT_SUM function can be used to return the digit sum of + a number. + +- #### app_sf: Add post-digit timer option to ReceiveSF. + The 't' option for ReceiveSF now allows for a timer since + the last digit received, in addition to the number-wide timeout. + +- #### app_dial: Allow fractional seconds for dial timeouts. + The answer and progress dial timeouts now have millisecond + precision, instead of having to be whole numbers. + +- #### chan_dahdi: Add DAHDI_CHANNEL function. + The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan. + +- #### app_queue.c: Add new global 'log_unpause_on_reason_change' + Add new global option 'log_unpause_on_reason_change' that + is default disabled. When enabled cause addition of UNPAUSE event on + every re-PAUSE with reason changed. + +- #### pbx_builtins: Allow custom tone for WaitExten. + The tone used while waiting for digits in WaitExten + can now be overridden by specifying an argument for the 'd' + option. + +- #### res_tonedetect: Add option for TONE_DETECT detection to auto stop. + The 'e' option for TONE_DETECT now allows detection to + be disabled automatically once the desired number of matches have + been fulfilled, which can help prevent race conditions in the + dialplan, since TONE_DETECT does not need to be disabled after + a hit. + +- #### sorcery: Prevent duplicate objects and ensure missing objects are created on update + Users relying on Sorcery multiple writable backends configurations + (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes + in sorcery.conf to ensure missing objects are recreated after temporary backend + failures. Default behavior remains unchanged unless explicitly enabled. + +- #### chan_websocket: Allow additional URI parameters to be added to the outgoing URI. + A new WebSocket channel driver option `v` has been added to the + Dial application that allows you to specify additional URI parameters on + outgoing connections. Run `core show application Dial` from the Asterisk CLI + to see how to use it. + +- #### app_chanspy: Add option to not automatically answer channel. + ChanSpy and ExtenSpy can now be configured to not + automatically answer the channel by using the 'N' option. + +- #### cel: Add STREAM_BEGIN, STREAM_END and DTMF event types. + Enabling the tracking of the + STREAM_BEGIN and the STREAM_END event + types in cel.conf will log media files and + music on hold played to each channel. + The STREAM_BEGIN event's extra field will + contain a JSON with the file details (path, + format and language), or the class name, in + case of music on hold is played. The DTMF + event's extra field will contain a JSON with + the digit and the duration in milliseconds. + +- #### res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM + Options are now available in the menuselect "Resource Modules" + category that allow you to enable the AES_192, AES_256 and AES_GCM + cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support + them but modern versions do. Previously, the only way to enable them was + to set the CFLAGS environment variable when running ./configure. + The default setting is to disable them preserving existing behavior. + +- #### cdr: add CANCEL dispostion in CDR + A new CDR option "canceldispositionenabled" has been added + that when set to true, the NO ANSWER disposition will be split into + two dispositions: CANCEL and NO ANSWER. The default value is 'no' + +- #### func_curl: Allow auth methods to be set. + The httpauth field in CURLOPT now allows the authentication + methods to be set. + +- #### Media over Websocket Channel Driver + A new channel driver "chan_websocket" is now available. It can + exchange media over both inbound and outbound websockets and will both frame + and re-time the media it receives. + See http://s.asterisk.net/mow for more information. + The ARI channels/externalMedia API now includes support for the +- #### res_stir_shaken.so: Handle X5U certificate chains. + The STIR/SHAKEN verification process will now load a full + certificate chain retrieved via the X5U URL instead of loading only + the end user cert. + +- #### res_stir_shaken: Add "ignore_sip_date_header" config option. + A new STIR/SHAKEN verification option "ignore_sip_date_header" has + been added that when set to true, will cause the verification process to + not consider a missing or invalid SIP "Date" header to be a failure. This + will make the IAT the sole "truth" for Date in the verification process. + The option can be set in the "verification" and "profile" sections of + stir_shaken.conf. + Also fixed a bug in the port match logic. + Resolves: #1251 + Resolves: #1271 + +- #### app_record: Add RECORDING_INFO function. + The RECORDING_INFO function can now be used + to retrieve the duration of a recording. + +- #### app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range + This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises + only for members whose current penalty is within the [min_penalty, max_penalty] range. + Members with lower or higher penalties are unaffected. + This behavior is backward-compatible with existing queue rule configurations. + +- #### res_odbc: cache_size option to limit the cached connections. + New cache_size option for res_odbc to on a per class basis limit the + number of cached connections. Please reference the sample configuration + for details. + +- #### res_odbc: cache_type option for res_odbc. + When using res_odbc it should be noted that back-end + connections to the underlying database can now be configured to re-use + the cached connections in a round-robin manner rather than repeatedly + re-using the same connection. This helps to keep connections alive, and + to purge dead connections from the system, thus more dynamically + adjusting to actual load. The downside is that one could keep too many + connections active for a longer time resulting in resource also begin + consumed on the database side. + +- #### ARI Outbound Websockets + Asterisk can now establish websocket sessions _to_ your ARI applications + as well as accepting websocket sessions _from_ them. + Full details: http://s.asterisk.net/ari-outbound-ws + +- #### res_websocket_client: Create common utilities for websocket clients. + A new module "res_websocket_client" and config file + "websocket_client.conf" have been added to support several upcoming new + capabilities that need common websocket client configuration. + +- #### asterisk.c: Add option to restrict shell access from remote consoles. + A new asterisk.conf option 'disable_remote_console_shell' has + been added that, when set, will prevent remote consoles from executing + shell commands using the '!' prefix. + Resolves: #GHSA-c7p6-7mvq-8jq2 + +- #### sig_analog: Add Call Waiting Deluxe support. + Call Waiting Deluxe can now be enabled for FXS channels + by enabling its corresponding option. + +- #### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout + A Dial timeout on POST /channels/{channelId}/dial will now result in a + CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously + no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing. + +- #### contrib: Add systemd service and timer files for malloc trim. + Service and timer files for systemd have been added to the + contrib/systemd/ directory. If you are experiencing memory issues, + install these files to have "malloc trim" periodically run on the + system. + +- #### Add log-caller-id-name option to log Caller ID Name in queue log + This patch adds a global configuration option, log-caller-id-name, to queues.conf + to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. + When log-caller-id-name=yes, the Caller ID name is included in the queue log, + Any '|' characters in the caller ID name will be replaced with '_'. + (provided it’s allowed by the existing log_restricted_caller_id rules). + When log-caller-id-name=no (the default), the Caller ID name is omitted. + +- #### asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. + In cli.conf, you can now define startup commands that run before + core initialization and before module initialization. + +- #### audiosocket: added support for DTMF frames + The AudioSocket protocol now forwards DTMF frames with + payload type 0x03. The payload is a 1-byte ascii representing the DTMF + digit (0-9,*,#...). + +- #### ari/pjsip: Make it possible to control transfers through ARI + Call transfers on the PJSIP channel can now be controlled by + ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) + dialplan function. + +- #### sig_analog: Add Last Number Redial feature. + Users can now redial the last number + called if the lastnumredial setting is set to yes. + Resolves: #437 + +- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms + The SHA-256 and SHA-512-256 algorithms are now available + for authentication as both a UAS and a UAC. + +- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016 + Bundled pjproject has been upgraded to 2.15.1. For more + information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.15.1 + +- #### res_pjsip: Add new AOR option "qualify_2xx_only" + The pjsip.conf AOR section now has a "qualify_2xx_only" + option that can be set so that only 2XX responses to OPTIONS requests + used to qualify a contact will mark the contact as available. + +- #### app_queue: allow dynamically adding a queue member in paused state. + use the p option of AddQueueMember() for paused member state. + Optionally, use the r(reason) option to specify a custom reason for the pause. + +- #### manager.c: Add Processed Call Count to CoreStatus output + The current processed call count is now returned as CoreProcessedCalls from the + CoreStatus AMI Action. + +- #### func_curl.c: Add additional CURL options for SSL requests + The following new configuration options are now available + in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost' + (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath' + (CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype' + (CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype', + (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the + libcurl documentation for more details. + +- #### res_stir_shaken: Allow sending Identity headers for unknown TNs + You can now set the "unknown_tn_attest_level" option + in the attestation and/or profile objects in stir_shaken.conf to + enable sending Identity headers for callerid TNs not explicitly + configured. + +- #### manager.c: Restrict ListCategories to the configuration directory. + The ListCategories AMI action now restricts files to the + configured configuration directory. + +- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" + The new "suppress_moh_on_sendonly" endpoint option + can be used to prevent playing MOH back to a caller if the remote + end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP. + +- #### app_mixmonitor: Add 'D' option for dual-channel audio. + The MixMonitor application now has a new 'D' option which + interleaves the recorded audio in the output frames. This allows for + stereo recording output with one channel being the transmitted audio and + the other being the received audio. The 't' and 't' options are + compatible with this. + +- #### manager.c: Restrict ModuleLoad to the configured modules directory. + The ModuleLoad AMI action now restricts modules to the + configured modules directory. + +- #### manager: Enhance event filtering for performance + You can now perform more granular filtering on events + in manager.conf using expressions like + `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/` + This is much more efficient than + `eventfilter = Event: Newchannel.*Channel: PJSIP/` + Full syntax guide is in configs/samples/manager.conf.sample. + +- #### db.c: Remove limit on family/key length + The `ast_db_*()` APIs have had the 253 byte limit on + "/family/key" removed and will now accept families and keys with a + total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This + affects the `DB*` dialplan applications, dialplan functions, + manager actions and `databse` CLI commands. Since the + media_cache also uses the `ast_db_*()` APIs, you can now store + resources with URIs longer than 253 bytes. + +- #### res_pjsip_notify: add dialplan application + A new dialplan application PJSIPNotify is now available + which can send SIP NOTIFY requests from the dialplan. + The pjsip send notify CLI command has also been enhanced to allow + sending NOTIFY messages to a specific channel. Syntax: + pjsip send notify