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
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.
Remove the deprecated 'rotatetimestamp' config option, as this
was deprecated by 'rotatestrategy' in 1.6 by commit
f5a14167f3.
Resolves: #1345
UpgradeNote: The deprecated rotatetimestamp option has been removed.
Use rotatestrategy instead.
The "no debug channel" command has been deprecated since
1.6 (commit 691363656f),
as it is replaced by "core set debug channel", which also
supports tab-completion on channels. Remove the redundant
command.
Resolves: #1343
UpgradeNote: The deprecated "no debug channel" command has
now been removed; use "core set debug channel" instead.
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
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'
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.
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
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
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
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
users.conf was deprecated in Asterisk 21 and is now being removed
for Asterisk 23, in accordance with the Asterisk deprecation policy.
This consists of:
* Removing integration with app_directory, app_voicemail, chan_dahdi,
chan_iax2, and AMI.
* users.conf was also partially used for res_phoneprov, and this remaining
functionality is consolidated to a separate phoneprov_users.conf,
used only by res_phoneprov.
Resolves: #1292
UpgradeNote: users.conf has been removed and all channel drivers must
be configured using their specific configuration files. The functionality
previously in users.conf for res_phoneprov is now in phoneprov_users.conf.
* Created chan_websocket which can exchange media over both inbound and
outbound websockets which the driver will frame and time.
See http://s.asterisk.net/mow for more information.
* res_http_websocket: Made defines for max message size public and converted
a few nuisance verbose messages to debugs.
* main/channel.c: Changed an obsolete nuisance error to a debug.
* ARI channels: Updated externalMedia to include chan_websocket as a supported
transport.
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.
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
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: #1276Resolves: #1279
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
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
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
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. `purge_sessions` iterates over that `session` (takes ref)
3. `purge_session` correctly de-referencess that session
4. `purge_session` re-evaluates the while() loop, taking a reference
5. `purge_session` exits (`n_max > 0` is false)
6. whatever allocated the `session` deallocates it, but a reference is
now lost since we exited the `while` loop before de-referencing.
7. since the destructor is never called, the session->last_ev->usecount
is never decremented, leading to events lingering in the queue
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.
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
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.
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().
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
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. 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.
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
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.
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:
* Added utilities to http.c:
* ast_get_http_method_from_string().
* ast_http_parse_post_form().
* Added utilities to json.c:
* ast_json_nvp_array_to_ast_variables().
* ast_variables_to_json_nvp_array().
* Added definitions for new events to carry REST responses.
* Created res/ari/ari_websocket_requests.c to house the new request handlers.
* Moved non-event specific code out of res/ari/resource_events.c into
res/ari/ari_websockets.c
* Refactored res/res_ari.c to move non-http code out of ast_ari_callback()
(which is http specific) and into ast_ari_invoke() so it can be shared
between both the http and websocket transports.
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/
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
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
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.
Commit 3cab4e7ab4 introduced a
regression by causing the wrong pointers to be used in certain
(more complex) cases. We now take care to ensure the exact
same pointers are used as before that commit, and simplify
by eliminating the unnecessary second for loop.
Resolves: #1147
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
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`
Issues:
* The bridging core allowed multiple bridges to be created with the same
unique bridgeId at the same time. Only the last bridge created with the
duplicate name was actually saved to the core bridges container.
* The bridging core was creating a stasis topic for the bridge and saving it
in the bridge->topic field but not increasing its reference count. In the
case where two bridges were created with the same uniqueid (which is also
the topic name), the second bridge would get the _existing_ topic the first
bridge created. When the first bridge was destroyed, it would take the
topic with it so when the second bridge attempted to publish a message to
it it either FRACKed or SEGVd.
* The bridge destructor, which also destroys the bridge topic, is run from the
bridge manager thread not the caller's thread. This makes it possible for
an ARI developer to create a new one with the same uniqueid believing the
old one was destroyed when, in fact, the old one's destructor hadn't
completed. This could cause the new bridge to get the old one's topic just
before the topic was destroyed. When the new bridge attempted to publish
a message on that topic, asterisk could either FRACK or SEGV.
* The ARI bridges resource also allowed multiple bridges to be created with
the same uniqueid but it kept the duplicate bridges in its app_bridges
container. This created a situation where if you added two bridges with
the same "bridge1" uniqueid, all operations on "bridge1" were performed on
the first bridge created and the second was basically orphaned. If you
attempted to delete what you thought was the second bridge, you actually
deleted the first one created.
Changes:
* A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if
a topic already exists for a bridge.
* `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in
resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check
if a bridge with the requested uniqueid already exists and will fail if it
does.
* `bridge_register()` in bridges.c now checks the core bridges container to
make sure a bridge doesn't already exist with the requested uniqueid.
Although most callers of `bridge_register()` will have already called
`bridge_base_init()`, which will now fail on duplicate bridges, there
is no guarantee of this so we must check again.
* The core bridges container allocation was changed to reject duplicate
uniqueids instead of silently replacing an existing one. This is a "belt
and suspenders" check.
* A global mutex was added to bridge.c to prevent concurrent calls to
`bridge_base_init()` and `bridge_register()`.
* Even though you can no longer create multiple bridges with the same uniqueid
at the same time, it's still possible that the bridge topic might be
destroyed while a second bridge with the same uniqueid was trying to use
it. To address this, the bridging core now increments the reference count
on bridge->topic when a bridge is created and decrements it when the
bridge is destroyed.
* `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges
container to make sure a bridge with the requested uniqueid doesn't already
exist. This may seem like overkill but there are so many entrypoints to
bridge creation that we need to be safe and catch issues as soon in the
process as possible.
* The stasis app_bridges container allocation was changed to reject duplicate
uniqueids instead of adding them. This is a "belt and suspenders" check.
* The `bridge show all` CLI command now shows the bridge name as well as the
bridge id.
* Response code 409 "Conflict" was added as a possible response from the ARI
bridge create resources to signal that a bridge with the requested uniqueid
already exists.
* Additional debugging was added to multiple bridging and stasis files.
Resolves: #211
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
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.
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_generator`s.
In passing, also note that `AST_AGENT_FD` isn't used either.
* Do a git blame on the embedded XML application or function element.
* 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 application or function
was defined.
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
* Do a git blame on the embedded XML managerEvent elements.
* 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 application or function
was defined.
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...
* The get_documentation awk script was exiting after it processed the first
DOCUMENTATION block it found in a file. We have at least 1 source file
with multiple DOCUMENTATION blocks so only the first one in them was being
processed. The awk script was changed to continue searching rather
than exiting after the first block.
* Fixing the awk script revealed an issue in logger.c where the third
DOCUMENTATION block contained a XML fragment that consisted only of
a managerEventInstance element that wasn't wrapped in a managerEvent
element. Since logger_doc.xml already existed, the remaining fragments
in logger.c were moved to it and properly organized.
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 <major>.<minor>.0.
* Sort and discard any <major>.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.
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
* channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element.
* main/bucket.c: Removed XML completely since the "bucket" and "file" objects
are internal only with no config file.
* main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf"
and should have been "acl.conf"
* res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element.
* res/res_http_media_cache.c: Fixed the configFile element name. It was
"http_media_cache.conf" and should have been "res_http_media_cache.conf".
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
