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
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.
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.
This change moves the PJSIP module from the threadpool API
to the taskpool API. PJSIP-specific implementations for
task usage have been removed and replaced with calls to
the optimized taskpool implementations instead. The need
for a pool of serializers has also been removed as
taskpool inherently provides this. The default settings
have also been changed to be more realistic for common
usage.
UpgradeNote: The threadpool_* options in pjsip.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 PJSIP. An alembic
upgrade script has been added to add these options to realtime
as well.
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: #1643Resolves: #1645
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
This conf file should be suffixed .sample so that make installs it
at compile time. Otherwise res_phoneprov complains at runtime as to
its absence and refuses to start.
Fixes: #1626
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 <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 <taskprocessor-name>
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
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
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.
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.
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
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
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
Convenience wrappers for showing modules at various support levels.
* module-list-core
* module-list-extended
* module-list-deprecated
Resolves: #1572
UserNote: Try "make module-list-deprecated" to see what modules
are on their way out the door.
Commit a46d5f9b76 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
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
* Fixed an issue with tarball-coredumps when asterisk was invoked without an
absolute path.
* Fixed an issue with gdb itself segfaulting when trying to get symbols from
separate debuginfo files. The command line arguments needed to be altered
such that the gdbinit files is loaded before anything else but the
`dump-asterisk` command is run after full initialization.
In the embedded gdbinit script:
* The extract_string_symbol function needed a `char *` cast to work properly.
* The s_strip function needed to be updated to continue to work with the
cpp_map_name_id channel storage backend.
* A new function was added to dump the channels when cpp_map_name_id was
used.
* The Channel object was updated to account for the new channel storage
backends
* The show_locks function was refactored to work correctly.
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(<seconds>)' (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.
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
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
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).
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.
* Added the chan_websocket.conf config file that allows setting which control
message format to use globally: "json" or "plain-text". "plain-text" is the
default for now to preserve existing behavior.
* Added a dialstring option `f(json|plain-text)` to allow the format to be
overridden on a call-by-call basis. Again, 'plain-text' is the default for
now to preserve existing behavior.
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: #1546Resolves: #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)"`.
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.
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.
* Using `==` with the POSIX sh `test` utility is UB.
* Switch back to using globs instead of using `$(find … | sort)`.
* Fix a missing redirect when checking for the OS type.
Resolves: #1554
Also...
* Refactored the verification datastore process so instead of having
a separate channel datastore for each verification result, there's only
one channel datastore with a vector of results.
* Refactored some log messages to include channel name and removed
some that would be redundant if a memory allocation failed.
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.
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
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
This change moves the PJSIP module from the threadpool API
to the taskpool API. PJSIP-specific implementations for
task usage have been removed and replaced with calls to
the optimized taskpool implementations instead. The need
for a pool of serializers has also been removed as
taskpool inherently provides this. The default settings
have also been changed to be more realistic for common
usage.
UpgradeNote: The threadpool_* options in pjsip.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 PJSIP. An alembic
upgrade script has been added to add these options to realtime
as well.
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
Commit 43bf8a4ded 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
