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
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.
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
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
Commit dc8e3eeaaf improved the debug log
messages in dsp.c. This makes two minor corrections to it:
* Properly guard an added log statement in a conditional.
* Don't add one to the hit count if there was no hit (however, we do
still want to do this for the case where this is one).
Resolves: #1496
When running "config show help <module>", if no XML documentation exists
for the specified module, "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
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.
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.
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. Synchronous tasks
3. Serializers for execution ordering of tasks
4. Growing/shrinking of number of taskprocessors in pool
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.
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
"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
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.
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. 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.
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.
* Added a new option to the WebSocket dial string to capture the additional
URI parameters.
* Added a new API ast_uri_verify_encoded() that verifies that a string
either doesn't need URI encoding or that it has already been encoded.
* Added a new API ast_websocket_client_add_uri_params() to add the params
to the client websocket session.
* Added XML documentation that will show up with `core show application Dial`
that shows how to use it.
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.
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:
* If we had a DSP hit, don't also emit the previous debug message that
was always logged. It is duplicated by the hit message, so this can
reduce the number of debug messages during detection by 50%.
* Include the hit count and required number of hits in the message so
on partial detections can be more easily troubleshot.
* Use debug level 9 for hits instead of 10, so we can focus on hits
without all the noise from the per-frame debug message.
* 1-index the hit count in the debug messages. On the first hit, it
currently logs '0', just as when we are not detecting anything,
which can be confusing.
Resolves: #1375
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.
* Fixed the rendering of examples so they are indented properly and changed
the color so they can be seen.
* There is now a line break before each option.
* Options are now printed on their own line with all option content indented
below them.
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.
- 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.
```
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.
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: #401Resolves: #1364
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.
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
* 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
