I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.

ASTERISK-27965

Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7

Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.

Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for.  If it finds it, it adds a "label"
attribute to the current sdp.

Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5

If a conference is ended very quickly after it was created (i.e., the
first user immediately hangs up) then the conference bridge and announcer
channels are not removed.

When a conference is created, the push_announcer() function is added to
the playback queue task processor and the conference object reference is
bumped.  If a conference is ended while the push_announcer() function is
still going then the ao2_cleanup(conference) at the end of
push_announcer() will call the destructor function -
destroy_conference_bridge().

The destroy_conference_bridge() function will then add the
hangup_playback() task to the playback queue and will wait for it to end.
Since it is already a current task of the playback queue it will wait
forever.

This patch makes the conference thread call push_announcer() directly.
This way the conference object reference bump is not needed.  Since the
playback queue task processor is only used by the conference thread
itself, there is no danger of trying to play announcements before the
announcer is pushed to the bridge.

ASTERISK-27870 #close

Change-Id: I947a50fb121422d90fd1816d643a54d75185a477

With the participant info code in app_confbridge, we were still
in the process of adding the channel to the bridge when trying to send
an in-dialog MESSAGE.  This caused 2 threads to grab the channel
blocking flag at the same time.  To mitigate this, the participant
info code was moved to confbridge_manager so it runs after all
channel/bridge actions have finished.

Change-Id: I228806ac153074f45e0b35d5236166e92e132abd

Add predial handler support to app_queue.  app_dial (ASTERISK_19548) and
app_originate (ASTERISK_26587) have the ability to execute predial
handlers on caller and callee channels.  This patch adds predial handlers
to app_queue and uses the same options as Dial and Originate (b and B).
The caller routine gets executed when the caller first enters the queue.
The callee routine gets executed for each queue member when they are about
to be called.

ASTERISK-27912

Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24

There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.

ASTERISK-27625

Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905

This patch changes the way asterisk polls output from mpg123, instead
of waiting for 10 seconds(when playing an http url) it now uses a
timeout of one second and iterates 10 times using this same timeout.

The main difference is that for every timeout asterisk receives it now
checks if mpg123 is still running before poll again.

ASTERISK-27752

Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620

ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc.  In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.

For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.

To control this behavior, the following options have been added to
confbridge.conf:

bridge_profile/enable_events:  This must be enabled on any bridge where
events are desired.

user_profile/send_events:  This must be set for a user profile to send
events.  Different user profiles connected to the same bridge can have
different settings.  This allows admins to get events but not normal
users for instance.

user_profile/echo_events:  In some cases, you might not want the user
triggering the event to get the event sent back to them.  To prevent it,
set this to false.

A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used.  This allows participant A's video
stream to appear as the same label to all other participants.

Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e

There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before.  Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.

Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9

When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.

ASTERISK-27877 #close

Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6

The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and
MeetmeTalkRequest AMI events were documented with sending out a Usernum
header when the User header was actually output.

* Change the online documentation to match reality.

ASTERISK-27873
ASTERISK-25261

Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39

Change-Id: Ieb8faf37dc765463ee5dbca1d1343242c756b1c7

Fix data-type mismatch between app_voicemail and database columns
exposed by new version of MariaDB

ASTERISK-27760

Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b

Correct the log warning message shown when ODBC voicemail
retrieve_file is called and there is a null value in the category
column.
A more meaningfull message is now written at debug level.

ASTERISK-27853

Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4

This fixes build warnings found by GCC 8.  In some cases format
truncation is intentional so the warning is just suppressed.

ASTERISK-27824 #close

Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84

Use AST_PBX_MAX_STACK to escape if we recurse 128 times.  This will
prevent crash if dialplan contains an include loop.  Log an error when
this occurs, at most one message per call to Macro() so we avoid logger
spam.

ASTERISK-26570 #close

Change-Id: I6c71b76998c31434391b150de055ae9a531e31da

Fixes a bug on the "confbridge show profile bridge" cli command
that showed "video_mode=no video" when video_mode was set
to "sfu"

ASTERISK-27418  #close

Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9

This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.

Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.

Support for configuring which behavior to use has been
added to app_confbridge.

ASTERISK-27804

Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66

SendText now accepts new channel variables that can be used
to override the To and From display names and set the Content-Type
of a message.  Since you can now set Content-Type, other text/*
content types are now valid.

Change-Id: I648b4574478119f95de09d9f08e9595831b02830

Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6

This change adds a configuration option to app_confbridge which can be
used to set the interval at which we will send a combined REMB (remote
estimated maximum bitrate) frame to sources of video. The bridging API
has also been extended slightly to allow setting this so bridge_softmix
can use it.

ASTERISK-27786

Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82

Add an option to make app_originate not wait for the created channel
to answer.

Change-Id: I7fc2facd77079abc6321f44e8bcd4e39298de2ae
Requested-by: Frederic Steinfels <fst@highdefinition.ch>
Signed-off-by: Russell Bryant <russell@russellbryant.net>

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