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${ noResults }
6690 Commits (6a2aad1d41398c77f2175084de4bcc02bba8611e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
7a0090325b |
manager.c: Add entries to Originate blacklist
Added Reload and DBdeltree to the list of dialplan application that can't be executed via the Originate manager action without also having write SYSTEM permissions. Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan functions that can't be executed via the Originate manager action without also having write SYSTEM permissions. If the Queue application is attempted to be run by the Originate manager action and an AGI parameter is specified in the app data, it'll be rejected unless the manager user has either the AGI or SYSTEM permissions. Resolves: #GHSA-c4cg-9275-6w44 |
1 year ago |
|
|
5c1822a481 |
app_voicemail_odbc: Allow audio to be kept on disk
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.
(cherry picked from commit
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1 year ago |
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963872daa7 |
asterisk.c: Fix sending incorrect messages to systemd notify
Send "RELOADING=1" instead of "RELOAD=1" to follow the format
expected by systemd (see sd_notify(3) man page).
Do not send STOPPING=1 in remote console mode:
attempting to execute "asterisk -rx" by the main process leads to
a warning if NotifyAccess=main (the default) or to a forced termination
if NotifyAccess=all.
(cherry picked from commit
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1 year ago |
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72db4a9530 |
res_http_websocket.c: Set hostname on client for certificate validation.
Additionally add a `assert()` to in the TLS client setup code to
ensure that hostname is set when it is supposed to be.
Fixes #433
(cherry picked from commit
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1 year ago |
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10c5aa65e8 |
tcptls/iostream: Add support for setting SNI on client TLS connections
If the hostname field of the ast_tcptls_session_args structure is
set (which it is for websocket client connections), that hostname
will now automatically be used in an SNI TLS extension in the client
hello.
Resolves: #713
UserNote: Secure websocket client connections now send SNI in
the TLS client hello.
(cherry picked from commit
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1 year ago |
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76f922cd1f |
rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
rtp_engine.c and stun.c were calling ast_register_cleanup which
is skipped if any loadable module can't be cleanly unloaded
when asterisk shuts down. Since this will always be the case,
their cleanup functions never get run. In a practical sense
this makes no difference since asterisk is shutting down but if
you're in development mode and trying to use the leak sanitizer,
the leaks from both of those modules clutter up the output.
(cherry picked from commit
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1 year ago |
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5e46323179 |
manager.c: Add missing parameters to Login documentation
* Added the AuthType and Key parameters for MD5 authentication.
* Added the Events parameter.
Resolves: #689
(cherry picked from commit
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1 year ago |
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4dcdded687 |
Fix incorrect application and function documentation references
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.
(cherry picked from commit
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1 year ago |
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a26faa9ac5 |
manager.c: Fix regression due to using wrong free function.
Commit |
2 years ago |
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e5f3c646e5 |
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.
(cherry picked from commit
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2 years ago |
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6415562a14 |
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.
(cherry picked from commit
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2 years ago |
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a669ccbc3d |
cel: add publish user event helper
Add a wrapper function around ast_cel_publish_event that
packs event and extras into a blob before publishing
Resolves:#330
(cherry picked from commit
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2 years ago |
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ed02f1c737 |
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.
(cherry picked from commit
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2 years ago |
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ddd78228f5 |
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options. In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.
OK so why would we want to include them? Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.
So...
* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
which tells make_buildopts_h to include the non-ABI-breaking
flags in buildopts.h as well as the ABI-breaking ones. The default
is disabled to preserve current behavior. As before though,
only the ABI-breaking flags appear in AST_BUILDOPTS and only
those are used to calculate AST_BUILDOPT_SUM.
A new AST_BUILDOPT_ALL define was created to capture all of the
flags.
* make_version_c was streamlined to use buildopts.h and also to
create asterisk_build_opts_all[] and ast_get_build_opts_all(void)
* "core show settings" now shows both AST_BUILDOPTS and
AST_BUILDOPTS_ALL.
UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
(cherry picked from commit
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2 years ago |
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b0ec10f9da |
manager: Tolerate stasis messages with no channel snapshot.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
(cherry picked from commit
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2 years ago |
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e14b1d7a3c |
audiohook: Unlock channel in mute if no audiohooks present.
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.
Resolves: #233
(cherry picked from commit
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2 years ago |
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11e772c642 |
manager.c: Prevent path traversal with GetConfig.
When using AMI GetConfig, it was possible to access files outside of the Asterisk configuration directory by using filenames with ".." and "./" even while live_dangerously was not enabled. This change resolves the full path and ensures we are still in the configuration directory before attempting to access the file. |
2 years ago |
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2915ce3cb9 |
utils: add lock timestamps for DEBUG_THREADS
Adds last locked and unlocked timestamps as well as a counter for the number of times the lock has been attempted (vs locked/unlocked) to debug output printed using the DEBUG_THREADS option. Resolves: #110 |
2 years ago |
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86bad8f550 |
build: Fix a few gcc 13 issues
* gcc 13 is now catching when a function is declared as returning an enum but defined as returning an int or vice versa. Fixed a few in app.h, loader.c, stasis_message.c. * gcc 13 is also now (incorrectly) complaining of dangling pointers when assigning a pointer to a local char array to a char *. Had to change that to an ast_alloca. Resolves: #155 |
2 years ago |
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0799c3d099 |
AMI: Add CoreShowChannelMap action.
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name and provides a list of all channels that are connected to that channel, following local channel connections as well. Resolves: #104 UserNote: New AMI action CoreShowChannelMap has been added. |
2 years ago |
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79e4c7e370 |
indications: logging changes
Increase verbosity to indicate failure due to missing country and to specify default on CLI dump Resolves: #89 |
2 years ago |
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5d225cfa1c |
cel: add local optimization begin event
The current AST_CEL_LOCAL_OPTIMIZE event is and has been triggered on a local optimization end to serve as a flag indicating the event occurred. This change adds a second AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail. Resolves: #52 UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired. UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion. |
2 years ago |
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730bf4a8b4 |
test.c: Fix counting of tests and add 2 new tests
The unit test XML output was counting all registered tests as "run" even when only a subset were actually requested to be run and the "failures" attribute was missing. * The "tests" attribute of the "testsuite" element in the output XML now reflects only the tests actually requested to be executed instead of all the tests registered. * The "failures" attribute was added to the "testsuite" element. Also added 2 new unit tests that just pass and fail to be used for CI testing. Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69 |
2 years ago |
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c8c528e9d3 |
res_mixmonitor: MixMonitorMute by MixMonitor ID
While it is possible to create multiple mixmonitor instances on a channel, it was not previously possible to mute individual instances. This change includes the ability to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via id. This id can be stored as a channel variable using the 'i' MixMonitor option. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor spy-type audiohooks on the channel. This is done via the new audiohook function: ast_audiohook_set_mute_all. ASTERISK-30464 Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98 |
2 years ago |
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d94555b4f4 |
bridge_builtin_features: add beep via touch variable
Add periodic beep option to one-touch recording by setting the touch variable TOUCH_MONITOR_BEEP or TOUCH_MIXMONITOR_BEEP to the desired interval in seconds. If the interval is less than 5 seconds, a minimum of 5 seconds will be imposed. If the interval is set to an invalid value, it will default to 15 seconds. A new test event PERIODIC_HOOK_ENABLED was added to the func_periodic_hook hook_on function to indicate when a hook is started. This is so we can test that the touch variable starts the hook as expected. ASTERISK-30446 Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661 |
2 years ago |
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ba51bb6089 |
cli: increase channel column width
For 'core show channels', the Channel name field is increased to 64 characters and the Location name field is increased to 32 characters. For 'core show channels verbose', the Channel name field is increased to 80 characters, the Context is increased to 24 characters and the Extension is increased to 24 characters. ASTERISK-30455 Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1 |
2 years ago |
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2285ffd5eb |
manager: prevent file access outside of config dir
Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404 |
3 years ago |
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f107e4fd87 |
audiohook: add directional awareness
Add enum to allow setting optional direction. If set to only one direction, only feed matching-direction frames to the associated slin factory. This prevents mangling the transcoder on non-mixed frames when the READ and WRITE frames would have otherwise required it. Also removes the need to mute or discard the un-wanted frames as they are no longer added in the first place. res_stasis_snoop is changed to use this addition to set direction on audiohook based on spy direction. If no direction is set, the ast_audiohook_init will init this enum to BOTH which maintains existing functionality. ASTERISK-30252 Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb |
3 years ago |
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3196ba7fcb |
res_geolocation: Fix segfault when there's an empty element
Fixed a segfault caused by var_list_from_loc_info() encountering an empty location info element. Fixed an issue in ast_strsep() where a value with only whitespace wasn't being preserved. Fixed an issue in ast_variable_list_from_quoted_string() where an empty value was considered a failure. ASTERISK-30215 Reported by: Dan Cropp Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a |
3 years ago |
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1bb1d3b5ef |
res_geolocation: Address user issues, remove complexity, plug leaks
* Added processing for the 'confidence' element. * Added documentation to some APIs. * removed a lot of complex code related to the very-off-nominal case of needing to process multiple location info sources. * Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes one eprofile instead of a datastore of multiples. * Plugged a huge leak in XML processing that arose from insufficient documentation by the libxml/libxslt authors. * Refactored stylesheets to be more efficient. * Renamed 'profile_action' to 'profile_precedence' to better reflect it's purpose. * Added the config option for 'allow_routing_use' which sets the value of the 'Geolocation-Routing' header. * Removed the GeolocProfileCreate and GeolocProfileDelete dialplan apps. * Changed the GEOLOC_PROFILE dialplan function as follows: * Removed the 'profile' argument. * Automatically create a profile if it doesn't exist. * Delete a profile if 'inheritable' is set to no. * Fixed various bugs and leaks * Updated Asterisk WiKi documentation. ASTERISK-30167 Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0 |
3 years ago |
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cc26b1d52b |
Geolocation: Base Asterisk Prereqs
* Added ast_variable_list_from_quoted_string() Parse a quoted string into an ast_variable list. * Added ast_str_substitute_variables_full2() Perform variable/function/expression substitution on an ast_str. * Added ast_strsep_quoted() Like ast_strsep except you can specify a specific quote character. Also added unit test. * Added ast_xml_find_child_element() Find a direct child element by name. * Added ast_xml_doc_dump_memory() Dump the specified document to a buffer * ast_datastore_free() now checks for a NULL datastore before attempting to destroy it. Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec |
3 years ago |
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4e604f03a7 |
xml.c, config,c: Add stylesheets and variable list string parsing
Added functions to open, close, and apply XML Stylesheets to XML documents. Although the presence of libxslt was already being checked by configure, it was only happening if xmldoc was enabled. Now it's checked regardless. Added ability to parse a string consisting of comma separated name/value pairs into an ast_variable list. The reverse of ast_variable_list_join(). Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e |
3 years ago |
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157a9e64c1 |
core: Config and XML tweaks needed for geolocation
Added:
Replace a variable in a list:
int ast_variable_list_replace_variable(struct ast_variable **head,
struct ast_variable *old, struct ast_variable *new);
Added test as well.
Create a "name=value" string from a variable list:
'name1="val1",name2="val2"', etc.
struct ast_str *ast_variable_list_join(
const struct ast_variable *head, const char *item_separator,
const char *name_value_separator, const char *quote_char,
struct ast_str **str);
Added test as well.
Allow the name of an XML element to be changed.
void ast_xml_set_name(struct ast_xml_node *node, const char *name);
Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb
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3 years ago |
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bdcaf87bf2 |
documentation: Adds missing default attributes.
The configObject tag contains a default attribute which allows the default value to be specified, if applicable. This allows for the default value to show up specially on the wiki in a way that is clear to users. There are a couple places in the tree where default values are included in the description as opposed to as attributes, which means these can't be parsed specially for the wiki. These are changed to use the attribute instead of being included in the text description. ASTERISK-29898 #close Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755 |
3 years ago |
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8d4a298cd4 |
GCC12: Fixes for 18+. state_id_by_topic comparing wrong value
GCC 12 caught an issue in state_id_by_topic where we were checking a pointer for NULL instead of the contents of the pointer for '\0'. ASTERISK-30044 Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e |
4 years ago |
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c0612ccc28 |
GCC12: Fixes for 16+
Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570 |
4 years ago |
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059eca1546 |
say.c: Prevent erroneous failures with 'say' family of functions.
A regression was introduced in ASTERISK~29531 that caused 'say'
functions to fail with file lists that would previously have
succeeded. This caused affected channels to hang up where previously
they would have continued.
We now explicitly check for the empty string to restore the previous
behavior.
ASTERISK-29859 #close
Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5
(cherry picked from commit
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4 years ago |
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5274bfdc07 |
say.c: Honor requests for DTMF interruption.
SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
and SayPhonetic all claim to allow DTMF interruption if the
SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
are failing to break out of a given 'say' application if DTMF actually
occurs.
ASTERISK-29816 #close
Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e
(cherry picked from commit
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4 years ago |
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c1129fdd8c |
bridge: Unlock channel during Local peer check.
It's not safe to keep the channel locked while locking
the peer Local channel, as it can result in a deadlock.
This change unlocks it during this time but keeps the
bridge locked to ensure nothing changes about the bridge.
ASTERISK-29821
Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6
(cherry picked from commit
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4 years ago |
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f946b92553 |
progdocs: Fix for Doxygen, the hidden parts.
ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b |
4 years ago |
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751bbf4b97 |
progdocs: Fix grouping for latest Doxygen.
Since Doxygen 1.8.16, a special comment block is required. Otherwise (pure C comment), the group command is ignored. Additionally, several unbalanced group commands were fixed. ASTERISK-29732 Change-Id: I4687857b9d56e6f44fd440b73af156691660202e |
4 years ago |
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04ac4fe509 |
config.c: Prevent UB in ast_realtime_require_field.
A backend's implementation of the realtime 'require' function may call va_arg() and then fail, leaving the va_list in an undefined state. Pass a copy of the va_list instead. ASTERISK-29771 #close Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4 |
4 years ago |
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70cdb0f9a8 |
app_voicemail: Refactor email generation functions
Refactors generic functions used for email generation into utils.c so that they can be used by multiple modules, including app_voicemail and app_minivm, to avoid code duplication. ASTERISK-29715 #close Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee |
4 years ago |
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31e385bebb |
xmldoc: Fix for Doxygen.
ASTERISK-29765 Change-Id: I654ba0debe8351038d4433716434a09370f04c9d |
4 years ago |
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ea941032ff |
astobj2.c: Fix core when ref_log enabled
In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj structure is freed, but is then referenced later if ref_log is enabled. The change is to store the obj->priv_data.options value locally and reference it instead of the value from the freed obj ASTERISK-29730 Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072 |
4 years ago |
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7d4e37a180 |
bridge: Deny full Local channel pair in bridge.
Local channels are made up of two pairs - the 1 and 2 sides. When a frame goes in one side, it comes out the other. Back and forth. When both halves are in a bridge this creates an infinite loop of frames. This change makes it so that bridging no longer allows both of these sides to exist in the same bridge. ASTERISK-29748 Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651 |
4 years ago |
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70b14f3eda |
rtp_engine: Add type field for JSON RTCP Report stasis messages
ASTERISK-29727 #close Change-Id: I2eca8aeb591cb63ac2238d08eab662367453cb82 |
4 years ago |
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b5962fe528 |
frame: Fix for Doxygen.
ASTERISK-29755 Change-Id: I8240013ec3db0669c0acf67e26bf6c9cbb5b72af |
4 years ago |
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e27b91d542 |
channel: Fix for Doxygen.
ASTERISK-29751 Change-Id: Ie04da5029c57ebee44733bdf05013156abe80176 |
4 years ago |
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6988386234 |
stasis: Fix for Doxygen.
ASTERISK-29750 Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9 |
4 years ago |