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
QUEUE_MEMBER_COUNT has been deprecated since at least 1.6,
for fully duplicating functionality available in the
QUEUE_MEMBER function; remove it now.
Resolves: #1341
UpgradeNote: The deprecated QUEUE_MEMBER_COUNT function
has been removed; use QUEUE_MEMBER(<queue>,logged) instead.
The already-deprecated "password" option for the AGENT function was
removed in commit d43b17a872 for
Asterisk 12, but the documentation for it wasn't removed then.
Resolves: #1321
Remove the deprecated maxmessage and minmessage options,
which have been superseded by maxsecs and minsecs since 1.6.
Also remove the deprecated 'cz' language option (deprecated
since 1.8.)
Resolves: #1298
UpgradeNote: The deprecated maxmessage and minmessage options
have been removed; use maxsecs and minsecs instead.
The deprecated 'cz' language has also been removed; use 'cs' instead.
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
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.
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.
This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2.
- Fix MORX message reception (from phone to Asterisk) in SMS protocol 2
- Fix MTTX message transmission (from Asterisk to phone) in 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.
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.
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
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.
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
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.
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,*,#...).
- Correct wait timeout logic in the dialplan application.
- Include server address in log messages for better traceability.
- Allow dialplan app to exit gracefully on hangup messages and socket closure.
- Optimize I/O by reducing redundant read()/write() operations.
Co-authored-by: Florent CHAUVEAU <florentch@pm.me>
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.
* 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.
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.
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.
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
Under some circumstances, the progress timeout feature added in commit
320c98eec8 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
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
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.
ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.
Resolves: #876
ChanSpy(${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
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:
* Tracing. The module is so complex that it was impossible to
figure out the path taken for various scenarios without the addition
of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
code that's not related to the functional change. Making this worse
was the fact that many "if" statements in this module didn't use
braces. Since the tracing macros add multiple statements, many "if"
statements had to be converted to use braces.
* Excessive use of PATH_MAX. Previous maintainers of this module
used PATH_MAX to allocate character arrays for filesystem paths
and SQL statements as though they cost nothing. In fact, PATH_MAX
is defined as 4096 bytes! Some functions had (and still have)
multiples of these. One function has 7. Given that the vast
majority of installations use the default spool directory path
`/var/spool/asterisk/voicemail`, the actual path length is usually
less than 80 bytes. That's over 4000 bytes wasted. It was the
same for SQL statement buffers. A 4K buffer for statement that
only needed 60 bytes. All of these PATH_MAX allocations in the
ODBC related code were changed to dynamically allocated buffers.
The rest will have to be addressed separately.
* Bug fixes. During the development of this feature, several
pre-existing ODBC related bugs were discovered and fixed. They
had to do with leaving orphaned files on disk, not preserving
original message ids when moving messages between folders,
not honoring the "formats" config parameter in certain circumstances,
etc.
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.
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.
This adds the RECORD_TIME variable to Record(),
which is set to the recording duration before
the application returns.
Resolves: #548
UpgradeNote: The RECORD_TIME variable now contains
the duration of Record() recordings in milliseconds.
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.
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.
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
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'.
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
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.
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.
