Sean Bright [Sat, 27 Jan 2024 14:46:27 +0000 (09:46 -0500)]
res_pjsip_session.c: Correctly format SDP connection addresses.
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.
Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:
```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```
And that value would be blindly copied into the outgoing SDP without
regard to its format.
Naveen Albert [Thu, 25 Jan 2024 17:44:29 +0000 (12:44 -0500)]
manager.c: Fix erroneous reloads in UpdateConfig.
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.
Sean Bright [Sun, 21 Jan 2024 16:40:19 +0000 (11:40 -0500)]
app_confbridge: Don't emit warnings on valid configurations.
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.
In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.
Mike Bradeen [Wed, 10 Jan 2024 16:47:24 +0000 (09:47 -0700)]
app_voicemail_odbc: remove macrocontext from voicemail_messages table
When app_macro was deprecated, the macrocontext column was removed from
the INSERT statement but the binds were not renumbered. This broke the
insert.
This change removes the macrocontext column via alembic and re-numbers
the existing columns in the INSERT.
Fixes: #527
UserNote: The fix requires removing the macrocontext column from the
voicemail_messages table in the voicemail database via alembic upgrade.
UpgradeNote: The fix requires that the voicemail database be upgraded via
alembic. Upgrading to the latest voicemail database via alembic will
remove the macrocontext column from the voicemail_messages table.
Naveen Albert [Fri, 10 Nov 2023 14:44:25 +0000 (09:44 -0500)]
chan_dahdi: Allow MWI to be manually toggled on channels.
This adds a CLI command to manually toggle the MWI status
of a channel, useful for troubleshooting or resetting
MWI devices, similar to the capabilities offered with
SIP messaging to manually control MWI status.
UserNote: The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Naveen Albert [Tue, 16 Jan 2024 21:58:22 +0000 (16:58 -0500)]
logger: Fix linking regression.
Commit 008731b0a4b96c4e6c340fff738cc12364985b64
caused a regression by resulting in logger.xml
being compiled and linked into the asterisk
binary in lieu of logger.c on certain platforms
if Asterisk was compiled in dev mode.
To fix this, we ensure the file has a unique
name without the extension. Most existing .xml
files have been named differently from any
.c files in the same directory or did not
pose this issue.
channels/pjsip/dialplan_functions.xml does not
pose this issue but is also being renamed
to adhere to this policy.
Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests. This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages. It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.
Naveen Albert [Sat, 23 Dec 2023 16:26:13 +0000 (16:26 +0000)]
app_if: Fix faulty EndIf branching.
This fixes faulty branching logic for the
EndIf application. Instead of computing
the next priority, which should be done
for false conditionals or ExitIf, we should
simply advance to the next priority.
Naveen Albert [Tue, 26 Dec 2023 17:14:11 +0000 (17:14 +0000)]
manager.c: Fix regression due to using wrong free function.
Commit 424be345639d75c6cb7d0bd2da5f0f407dbd0bd5 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.
Naveen Albert [Tue, 5 Dec 2023 16:36:47 +0000 (11:36 -0500)]
manager.c: Improve clarity of "manager show connected".
Improve the "manager show connected" CLI command
to clarify that the last two columns are permissions
related, not counts, and use sufficient widths
to consistently display these values.
Although `make_xml_documentation`'s `print_dependencies` command was
corrected by the previous fix (#461) for #142, the `create_xml` was
not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.
Naveen Albert [Thu, 9 Nov 2023 21:26:46 +0000 (16:26 -0500)]
general: Fix broken links.
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36a70e81fe373a3d19132c43adf3f74b.
app_dial: Add option "j" to preserve initial stream topology of caller
Resolves: #462
UserNote: The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
Sean Bright [Sat, 2 Dec 2023 16:13:31 +0000 (11:13 -0500)]
pbx_config.c: Don't crash when unloading module.
`pbx_config` subscribes to manager events to capture the `FullyBooted`
event but fails to unsubscribe if the module is loaded after that
event fires. If the module is unloaded, a crash occurs the next time a
manager event is raised.
We now unsubscribe when the module is unloaded if we haven't already
unsubscribed.
George Joseph [Sun, 12 Nov 2023 00:40:10 +0000 (17:40 -0700)]
ast_coredumper: Increase reliability
Instead of searching for the asterisk binary and the modules in the
filesystem, we now get their locations, along with libdir, from
the coredump itself...
For the binary, we can use `gdb -c <coredump> ... "info proc exe"`.
gdb can print this even without having the executable and symbols.
Once we have the binary, we can get the location of the modules with
`gdb ... "print ast_config_AST_MODULE_DIR`
If there was no result then either it's not an asterisk coredump
or there were no symbols loaded. Either way, it's not usable.
For libdir, we now run "strings" on the note0 section of the
coredump (which has the shared library -> memory address xref) and
search for "libasteriskssl|libasteriskpj", then take the dirname.
Since we're now getting everything from the coredump, it has to be
correct as long as we're not crossing namespace boundaries like
running asterisk in a docker container but trying to run
ast_coredumper from the host using a shared file system (which you
shouldn't be doing).
There is still a case for using --asterisk-bin and/or --libdir: If
you've updated asterisk since the coredump was taken, the binary,
libraries and modules won't match the coredump which will render it
useless. If you can restore or rebuild the original files that
match the coredump and place them in a temporary directory, you can
use --asterisk-bin, --libdir, and a new --moddir option to point to
them and they'll be correctly captured in a tarball created
with --tarball-coredumps. If you also use --tarball-config, you can
use a new --etcdir option to point to what normally would be the
/etc/asterisk directory.
Sean Bright [Fri, 1 Dec 2023 17:03:29 +0000 (12:03 -0500)]
logger.c: Move LOG_GROUP documentation to dedicated XML file.
The `get_documentation` awk script will only extract the first
DOCUMENTATION block that it finds in a given file. This is by design
(9bc2127) to prevent AMI event documentation from being pulled in to
the core.xml documentation file.
Because of this, the `LOG_GROUP` documentation added in 89709e2 was
not being properly extracted and was missing fom the resulting XML
documentation file. This commit moves the `LOG_GROUP` documentation to
a separate `logger.xml` file.
res_odbc.c: Allow concurrent access to request odbc connections
There are valid scenarios where res_odbc's connection pool might have some dead
or stuck connections while others are healthy (imagine network
elements/firewalls/routers silently timing out connections to a single DB and a
single IP address, or a heterogeneous connection pool connected to potentially
multiple IPs/instances of a replicated DB using a DNS front end for load
balancing and one replica fails).
In order to time out those unhealthy connections without blocking access to
other parts of Asterisk that may attempt access to the connection pool, it would
be beneficial to not lock/block access around the entire pool in
_ast_odbc_request_obj2 while doing potentially blocking operations on connection
pool objects such as the connection_dead() test, odbc_obj_connect(), or by
dereferencing a struct odbc_obj for the last time and triggering a
odbc_obj_disconnect().
This would facilitate much quicker and concurrent timeout of dead connections
via the connection_dead() test, which could block potentially for a long period
of time depending on odbc.ini or other odbc connector specific timeout settings.
This also would make rapid failover (in the clustered DB scenario) much quicker.
This patch changes the locking in _ast_odbc_request_obj2() to not lock around
odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
lock around truly shared, non-immutable state like the connection_cnt member and
the connections list on struct odbc_class.
Sean Bright [Wed, 22 Nov 2023 19:25:19 +0000 (14:25 -0500)]
config.c: Log #exec include failures.
If the script referenced by `#exec` does not exist, writes anything to
stderr, or exits abnormally or with a non-zero exit status, we log
that to Asterisk's error logging channel.
Additionally, write out a warning if the script produces no output.
Resequencing is a process that occurs when we open a voicemail folder
and discover that there are gaps between messages (e.g. `msg0000.txt`
is missing but `msg0001.txt` exists). Resequencing involves shifting
the existing messages down so we end up with a sequential list of
messages.
Currently, this process stops after reaching a threshold based on the
message limit (`maxmsg`) configured on the current folder. However, if
`maxmsg` is lowered when a voicemail folder contains more than
`maxmsg + 10` messages, resequencing will not run completely leaving
the mailbox in an inconsistent state.
We now resequence up to the maximum number of messages permitted by
`app_voicemail` (currently hard-coded at 9999 messages).
Naveen Albert [Fri, 24 Nov 2023 13:06:58 +0000 (08:06 -0500)]
sig_analog: Fix channel leak when mwimonitor is enabled.
When mwimonitor=yes is enabled for an FXO port,
the do_monitor thread will launch mwi_thread if it thinks
there could be MWI on an FXO channel, due to the noise
threshold being satisfied. This, in turns, calls
analog_ss_thread_start in sig_analog. However, unlike
all other instances where __analog_ss_thread is called
in sig_analog, this call path does not properly set
pvt->ss_astchan to the Asterisk channel, which means
that the Asterisk channel is NULL when __analog_ss_thread
starts executing. As a result, the thread exits and the
channel is never properly cleaned up by calling ast_hangup.
This caused issues with do_monitor on incoming calls,
as it would think the channel was still owned even while
receiving events, leading to an infinite barrage of
warning messages; additionally, the channel would persist
improperly.
To fix this, the assignment is added to the call path
where it is missing (which is only used for mwi_thread).
A warning message is also added since previously there
was no indication that __analog_ss_thread was exiting
abnormally. This resolves both the channel leak and the
condition that led to the warning messages.
Certain channel options are not set anywhere or
exposed in any way to users, making them unusable.
This exposes some of these options which make sense
for users to manipulate at runtime.
Sean Bright [Tue, 7 Nov 2023 20:03:53 +0000 (15:03 -0500)]
app.c: Allow ampersands in playback lists to be escaped.
Any function or application that accepts a `&`-separated list of
filenames can now include a literal `&` in a filename by wrapping the
entire filename in single quotes, e.g.:
UpgradeNote: Ampersands in URLs passed to the `Playback()`,
`Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
`Queue()` applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the `CONFBRIDGE` dialplan function, or configuring various
features in `confbridge.conf` and `queues.conf`.
Sean Bright [Thu, 9 Nov 2023 14:11:33 +0000 (09:11 -0500)]
live_ast: Add astcachedir to generated asterisk.conf.
`astcachedir` (added in b0842713) was not added to `live_ast` so
continued to point to the system `/var/cache` directory instead of the
one in the live environment.
app_followme.c: Grab reference on nativeformats before using it
Fixes a crash due to a lack of proper reference on the nativeformats
object before passing it into ast_request(). Also found potentially
similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c
Naveen Albert [Thu, 9 Nov 2023 15:12:15 +0000 (10:12 -0500)]
configs: Improve documentation for bandwidth in iax.conf.
This improves the documentation for the bandwidth setting
in iax.conf by making it clearer what the ramifications
of this setting are. It also changes the sample default
from low to high, since only high is compatible with good
codecs that people will want to use in the vast majority
of cases, and this is a common gotcha that trips up new users.
Naveen Albert [Wed, 9 Aug 2023 22:12:55 +0000 (22:12 +0000)]
logger: Add channel-based filtering.
This adds the ability to filter console
logging by channel or groups of channels.
This can be useful on busy systems where
an administrator would like to analyze certain
calls in detail. A dialplan function is also
included for the purpose of assigning a channel
to a group (e.g. by tenant, or some other metric).
ASTERISK-30483 #close
Resolves: #242
UserNote: The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
George Joseph [Tue, 7 Nov 2023 23:24:37 +0000 (16:24 -0700)]
codec_ilbc: Disable system ilbc if version >= 3.0.0
Fedora 37 started shipping ilbc 3.0.4 which we don't yet support.
configure.ac now checks the system for "libilbc < 3" instead of
just "libilbc". If true, the system version of ilbc will be used.
If not, the version included at codecs/ilbc will be used.
George Joseph [Tue, 31 Oct 2023 21:08:14 +0000 (15:08 -0600)]
chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
Sean Bright [Mon, 6 Nov 2023 16:45:33 +0000 (11:45 -0500)]
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE
in a frame was one that may not have any data - such as the CALLTOKEN
IE in an NEW request - it was not getting displayed.
Naveen Albert [Wed, 27 Sep 2023 12:49:07 +0000 (08:49 -0400)]
app_directory: Add ADSI support to Directory.
This adds optional ADSI support to the Directory
application, which allows callers with ADSI CPE
to navigate the Directory system significantly
faster than is possible using the audio prompts.
Callers can see the directory name (and optionally
extension) on their screenphone and confirm or
reject a match immediately rather than waiting
for it to be spelled out, enhancing usability.
Naveen Albert [Wed, 9 Aug 2023 22:57:41 +0000 (22:57 +0000)]
core_local: Fix local channel parsing with slashes.
Currently, trying to call a Local channel with a slash
in the extension will fail due to the parsing of characters
after such a slash as being dial modifiers. Additionally,
core_local is inconsistent and incomplete with
its parsing of Local dial strings in that sometimes it
uses the first slash and at other times it uses the last.
For instance, something like DAHDI/5 or PJSIP/device
is a perfectly usable extension in the dialplan, but Local
channels in particular prevent these from being called.
This creates inconsistent behavior for users, since using
a slash in an extension is perfectly acceptable, and using
a Goto to accomplish this works fine, but if specified
through a Local channel, the parsing prevents this.
This fixes this by explicitly parsing options from the
last slash in the extension, rather than the first one,
which doesn't cause an issue for extensions with slashes.
Naveen Albert [Mon, 30 Oct 2023 13:27:20 +0000 (09:27 -0400)]
app_voicemail: Add AMI event for mailbox PIN changes.
This adds an AMI event that is emitted whenever a
mailbox password is successfully changed, allowing
AMI consumers to process these.
UserNote: The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
In simple_bridge_join, we were sending topology change requests
even when the new and old topologies were the same. In some
circumstances, this can cause unnecessary re-invites and even
a re-invite flood. We now suppress those.
Mike Bradeen [Thu, 7 Sep 2023 17:21:32 +0000 (11:21 -0600)]
res_speech: allow speech to translate input channel
* Allow res_speech to translate the input channel if the
format is translatable to a format suppored by the
speech provider.
Resolves: #129
UserNote: res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
George Joseph [Thu, 19 Oct 2023 13:40:26 +0000 (07:40 -0600)]
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
Commit f66f77f last year prevents the res_pjsip_exten_state and
res_pjsip_mwi modules from unloading due to possible pjproject
asserts if the modules are reloaded. A side effect of the
implementation is that the taskprocessors these modules use aren't
being released. When asterisk is doing a graceful shutdown, it
waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all
taskprocessors to stop but since those 2 modules don't release
theirs, the shutdown hangs for that amount of time.
This change allows the modules to be unloaded and their resources to
be released when ast_shutdown_final is true.
sungtae kim [Fri, 22 Sep 2023 17:32:43 +0000 (02:32 +0900)]
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
This commit introduces an extension to the endpoint and relevant
resource sizes for PJSIP, transitioning from its current 40-character
constraint to a more versatile 255-character capacity. This enhancement
significantly overcomes limitations related to domain qualification and
practical usage, ultimately delivering improved functionality. In
addition, it includes adjustments to accommodate the expanded realm size
within the ARI, specifically enhancing the maximum realm length.
Resolves: #345
UserNote: With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
UpgradeNote: As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
Mike Bradeen [Mon, 2 Oct 2023 18:35:32 +0000 (12:35 -0600)]
res_stasis: signal when new command is queued
res_statsis's app loop sleeps for up to .2s waiting on input
to a channel before re-checking the command queue. This can
cause delays between channel setup and bridge.
This change is to send a SIGURG on the sleeping thread when
a new command is enqueued. This exits the sleeping thread out
of the ast_waitfor() call triggering the new command being
processed on the channel immediately.
Resolves: #362
UserNote: Call setup times should be significantly improved
when using ARI.
George Joseph [Thu, 5 Oct 2023 14:18:17 +0000 (08:18 -0600)]
logger.h: Add ability to change the prefix on SCOPE_TRACE output
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else. Full documentation in logger.h.
George Joseph [Thu, 21 Sep 2023 17:12:31 +0000 (11:12 -0600)]
Add libjwt to third-party
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.
A few tweaks were also made to the third-party infrastructure as
a whole. The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.
Mike Bradeen [Tue, 26 Sep 2023 19:19:27 +0000 (13:19 -0600)]
res_pjsip: update qualify_timeout documentation with DNS note
The documentation on qualify_timeout does not explicitly state that the timeout
includes any time required to perform any needed DNS queries on the endpoint.
If the OPTIONS response is delayed due to the DNS query, it can still render an
endpoint as Unreachable if the net time is enough for qualify_timeout to expire.
Naveen Albert [Mon, 4 Sep 2023 11:58:23 +0000 (07:58 -0400)]
chan_dahdi: Clarify scope of callgroup/pickupgroup.
Internally, chan_dahdi only applies callgroup and
pickupgroup to FXO signalled channels, but this is
not documented anywhere. This is now documented in
the sample config, and a warning is emitted if a
user tries configuring these settings for channel
types that do not support these settings, since they
will not have any effect.
Mike Bradeen [Thu, 21 Sep 2023 18:54:45 +0000 (12:54 -0600)]
res_speech_aeap: add aeap error handling
res_speech_aeap previously did not register an error handler
with aeap, so it was not notified of a disconnect. This resulted
in SpeechBackground never exiting upon a websocket disconnect.
Naveen Albert [Wed, 27 Sep 2023 12:28:50 +0000 (08:28 -0400)]
app_voicemail: Disable ADSI if unavailable.
If ADSI is available on a channel, app_voicemail will repeatedly
try to use ADSI, even if there is no CPE that supports it. This
leads to many unnecessary delays during the session. If ADSI is
available but ADSI setup fails, we now disable it to prevent
further attempts to use ADSI during the session.
Some providers require a multiple of 20 for the maxptime or fail to complete calls,
e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.
George Joseph [Wed, 13 Sep 2023 15:18:04 +0000 (09:18 -0600)]
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
George Joseph [Fri, 15 Sep 2023 17:19:18 +0000 (11:19 -0600)]
asterisk.c: Use the euid's home directory to read/write cli history
The CLI .asterisk_history file is read from/written to the directory
specified by the HOME environment variable. If the root user starts
asterisk with the -U/-G options, or with runuser/rungroup set in
asterisk.conf, the asterisk process is started as root but then it
calls setuid/setgid to set the new user/group. This does NOT reset
the HOME environment variable to the new user's home directory
though so it's still left as "/root". In this case, the new user
will almost certainly NOT have access to read from or write to the
history file.
* Added function process_histfile() which calls
getpwuid(geteuid()) and uses pw->dir as the home directory
instead of the HOME environment variable.
* ast_el_read_default_histfile() and ast_el_write_default_histfile()
have been modified to use the new process_histfile()
function.
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
From the gdb information, ast_websocket_read reads a message successfully,
then transport_read is called in the serializer. During execution of pjsip_transport_down,
ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop.
After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages.
This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue.
In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.
Naveen Albert [Sat, 9 Sep 2023 15:24:37 +0000 (11:24 -0400)]
chan_console: Fix deadlock caused by unclean thread exit.
To terminate a console channel, stop_stream causes pthread_cancel
to make stream_monitor exit. However, commit 5b8fea93d106332bc0faa4b7fa8a6ea71e546cac
added locking to this function which results in deadlock due to
the stream_monitor thread being killed while it's holding the pvt lock.
To resolve this, a flag is now set and read to indicate abort, so
the use of pthread_cancel and pthread_kill can be avoided altogether.
George Joseph [Mon, 11 Sep 2023 14:27:41 +0000 (08:27 -0600)]
file.c: Add ability to search custom dir for sounds
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory. If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory. For performance
reasons, the "sounds_search_custom_dir" defaults to "false".
Resolves: #315
UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.
George Joseph [Fri, 15 Sep 2023 16:17:11 +0000 (10:17 -0600)]
res_pjsip_pubsub: Add body_type to test_handler for unit tests
The ast_sip_subscription_handler "test_handler" used for the unit
tests didn't set "body_type" so the NULL value was causing
a SEGV in build_subscription_tree(). It's now set to "".
projects / thirdparty / asterisk.git / log
