This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
Add a reference to the calling module when it is active to protect
access to datastore->info. Remove module references done by
func_periodic_hook as the datastore now handles it.
ASTERISK-25128 #close
Change-Id: I8357a3711e77591d0d1dd8ab4211a7eedd782c89
Once an Optional API module is loaded it should stay loaded. Unloading
an optional API module runs the risk of a crash if something else is
using it. This patch causes all optional API providers to tell the
module loader not to unload except at shutdown.
ASTERISK-27389
Change-Id: Ia07786fe655681aec49cc8d3d96e06483b11f5e6
In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
These modules save a pointer to the context they create on load, and
use that pointer to destroy the context at unload. It is not safe
to save this pointer, it is replaced during load of pbx_config,
pbx_lua or pbx_ael.
This change causes the modules to pass NULL to ast_context_destroy,
a safer way to perform the unregistration since it does not use
a pointer that could become invalid.
ASTERISK-25085 #close
Reported by: Corey Farrell
Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
For some odd reason, loading app_mixmonitor was fine, but res_monitor was not.
This patch fixes a set of issues related to func_periodic_hook exporting the
beep functions that gets res_monitor working again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add an option to enable a periodic beep to be played into a call if it
is being recorded. If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval. This option is provided for both Monitor() and
MixMonitor().
Review: https://reviewboard.asterisk.org/r/3424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous code left one error path where the module would be unref'd twice
instead of once. It was done once in the error handling block, and again inside
of datastore destruction.
Now the module ref is only released in the datastore destructor and only acquired
when the datastore has been successfully allocated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This module lacked necessary module ref count incrementing and decrementing
when used. This patch adds it. There's already a datastore used, so doing the
ref counting along with the lifetime of the datastore provides a convenient
place to do it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This module makes use of some existing Asterisk components. app_chanspy was
already listed as a dependency. There are a few function modules used, as
well, so list them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically. The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded. The implementation is much
more generic though and could be used for many other things.
The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.
The other important bit of the implementation is how it figures out
when to trigger the beep playback. This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way. It's a convenient way to get a callback and check if it's time
to kick off another beep. It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.
Review: https://reviewboard.asterisk.org/r/3362/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Corey Farrell
Sean Bright
Rodrigo Ramírez Norambuena
Matt Jordan
Mark Michelson
Russell Bryant