Asterisk does not need the development package of libltdl, because it does not
use any symbol of -lltdl directly. Instead, it uses the runtime package via the
shared library -lodbc. On the supported platforms, that shared library declares
its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have
failed.

ASTERISK-27745

Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba

Certain applications (e.g. door-phone) require that also video is transmitted
before a call is accepted.

Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e

When app_voicemail calls ast_test_suite_notify with the results of
a user keypress, it formats the keypress as '%c'.  If the user hung up
or some other error occurrs, the result of the keypress is a non
printable character.  This ultimately causes json_vpack_ex to think
it's being passed a non utf-8 string and return an error.

* Keypress results passed to ast_test_suite_notify are now checked with
  isprint() and a '?' is substituted if the check fails.

Change-Id: I78ee188916bbac840f3d03f40201b692347ea865

* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl

These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.

Some of these modules are still initialized or shutdown from outside the
module loader.  logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).

Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f

The menuselect comment was updated to deprecate these modules but the
AST_MODULE_INFO block at the end of file was missed.

ASTERISK-27671

Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839

Checking option_debug directly is incorrect as it ignores file/module
specific debug settings.  This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.

Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0

Fixed wrong voicemail mailbox reference for Action: VoicemailUsersList.

ASTERISK-27703

Change-Id: Ie6578ad80bba2bfaf34b84f0be978f59045ce6cd

astosp.h is leftover from when logic was split between app_osplookup and
res_osp.  All logic was moved into app_osplookup by 109737eb in 2006,
but astosp.h remained.  This moves the remaining defines into
app_osplookup and deletes astosp.h.

Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd

Between Asterisk 11 and Asterisk 13 there was a significant increase
in the number of AST_FRAME_NULL frames being processed by app_amd.c's
main loop. Each AST_FRAME_NULL frame was being counted as 100ms
towards the total time and silence. This may have been accurate
when app_amd.c was orginally added, but it is not in Asterisk 13.
As such the total analysis time and silence calculations were way
off effectively breaking app_amd.c

* Additional debug messages were added
* AST_FRAME_NULL are now ignored

ASTERISK-27610

Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9

* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).

This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.

ASTERISK-27671 #close

Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49

* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers.  The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped.  However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".

ASTERISK-27651

Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977

ASTERISK-27651

Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34

The dsp_talking_threshold does not represent time in milliseconds.  It
represents the average magnitude per sample in the audio packets.  This is
what the DSP uses to determine if a packet is silence or talking/noise.

Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443

clang 4.0 warned about this.

ASTERISK-27635

Change-Id: I213f230607d7fbe97c0f5f2d60da9cbf5a2d8231

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

I've audited all modules that include any header which includes
asterisk/optional_api.h.  All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.

In addition ARI dependency declarations have been reworked.  Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.

Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606

This patch adds the ability to configure a prompt which will be read
to the "winner" who pressed 1 (or the configured value) and received
the call.

ASTERISK-24372 #close

Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272

* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.

Still need to investigate dependencies among modules I cannot compile.

Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf

GCC 7 warned about this.

ASTERISK-27578

Change-Id: I4a00458dbe9b575ef04338b6a7852272745e1552

The check for last_user == NULL needs to happen before we dereference
the variable, previously it was possible for us to check flags of a NULL
last_user.

Change-Id: I274f737aa8af9d2d53e4a78cdd7ad57561003945

clang 5.0 warned about this.

ASTERISK-27557

Change-Id: I7cceaa88e147cbdf81a3a7beec5c1c20210fa41e

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Fix error where input/output file descriptors would be closed multiple
times.

Change-Id: Iba5140b60cb7de79e3d5d92be3c256947aa99da9

Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh

ASTERISK-24198 #close

Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31

Change-Id: If5efaddcf735ff7d17e55c36cc1388946cee9e0f

* mwi_sub_event_cb: mwist leaked on separate_mailbox failure.
* add_email_attachment: A reference to sox_gain_tmpdir was used
  after the storage was out of scope.

Change-Id: I6282c542ff7b82fa091177a912d11234a8b00a30

This patch adds the ability to set the wrapuptime on the queue member
config.

When the option is set the wrapuptime on the queue member is used instead
of the queue's wrapuptime.

ASTERISK-27483 #close

Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902

The Local channel has never supported app_transfer
from what I can see so remove it from the documentation.

ASTERISK-25649

Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9

Remove nearly all use of regex from ACO users.  Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
  callers use simple prefix based regex.  I haven't decided the best
  way to fix this in both 13/15 and master.

Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b

Currently, to figure out specified voicemail's status, there's only one
way to do it, which is use a VoicemailUserEntry AMI message.
But it consumed it too much resource(it check everything).
So, added new AMI action.

ASTERISK-27470

Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7

The approach with having a single global subscription to all extension
state changes has one issue: dynamically created hints don't have any
watchers and are therefore garbage collected on the first dialplan
reload.

This change creates a state subscription for every queue member with a
hint as state_interface, thus increasing the count of watches for
hints, so they are not destroyed prematurely anymore.

There are 2 side effects:
1. The state change callback in app_queue is not executed when
   there are no members referring to the extension.
2. The callback is called multiple times for the same hint if it's
   associated with more than one queue member.

Reported by: Steven T. Wheeler

ASTERISK-18411 #close

Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89

There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.

Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7

Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's
no OldMessageCount info for default.
To check the OldMessageCount info, it required IMAP_STORAGE define, but this is
not correct.
Added OldMessageCount item as a default.

ASTERISK-27456

Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7

Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.

Change-Id: I0123258eafce324249433a69df15a85cc16e509f

Declare 'res' initialized to -1 to deal with earlier error paths that
could cause 'res' to be returned uninitialized.

Change-Id: I8ac2a5755bf4174d89ef893e924c940f702b104e

We've been calling pbx_builtin_setvar_helper to set the
RECORD_STATUS variable before actually closing the recorded file.
If a client is watching VarSet events and tries to do something with
the file when a RECORD_STATUS event is seen, they might attempt to
do so while the file it's still open.

We now delay calling pbx_builtin_setvar_helper until after we close
the file.

ASTERISK-27423

Change-Id: I7fe9de99953e46b4bafa2b38cf151fe8f6488254

When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it.  The most likely way it can return failure is for an
allocation failure.  If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.

* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure.  That is much more useful than either an uninitialized pointer
or a pointer that has already been freed.  Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.

* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().

* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.

Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23

This adds menuselect dependencies for modules that use symbols of other
modules.

ASTERISK-27390

Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385

* Stop using ast_module_helper to check if a module is loaded, use
  ast_module_check instead (app_confbridge and app_meetme).
* Stop ast_module_helper from listing reload classes when needsreload
  was not requested.

ASTERISK-27378

Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239

Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'

ASTERISK-27377 #close

Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930