Change-Id: I1946509f518961d23fb21229d91676ee3e441921

Declining the queue_member_status_type stasis message in stasis.conf
causes these messages to leak json objects.

* Add missing ast_json_unref() if the type is NULL in
queue_publish_member_blob().

ASTERISK-28084

Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc

The first attempt at publishing confbridge events to participants
involved publishing them at the same time stasis events were
created.  This caused issues with bridge and channel locks.  The
second attempt involved publishing them when the stasis events
were received by the code that published the confbridge AMI events.
This caused timing issues because, depending on resources available,
the event could be received before channels actually joined the
bridge and would therefore fail to send messages to the participant.

This attempt reverts to the original mechanism with one exception.
The join and leave events are published via bridge join and leave
hooks.  This guarantees the states of the channels and bridge and
provides deterministic timing for event publishing.

Change-Id: I2660074f8a30a5224cb953d5e047ee84484a9036

This issue related to setting of holdtime, announcements, member delays.
It works well if we set the member delays to "0" and no announcements
and no holdtime.This issue will happen if we set member delays to "1",
"2"... or announcements or holdtime and hangs up the call during
processing it.

And here is the reason:
(At the step of answering a phone.)
It takes care any holdtime, announcements, member delays,
or other options after a call has been answered if it exists.

Normally, After the call has been aswered,
and we wait for the processing one of the cases of the member delays
or hold time or announcements finished, "if (ast_check_hangup(peer))"
will be not executed, then queue will be updated at update_queue().
Here, pending member will be removed.

However, after the call has been aswered,
if we hangs up the call during one of the cases of the member delays
or hold time or announcements, "if (ast_check_hangup(peer))"
will be executed.
outgoing = NULL and at hangupcalls, pending members will not be removed.

* This fixed patch will remove the pending member from container
before hanging up the call with outgoing is NULL.

ASTERISK-27920

Reported by: Cao Minh Hiep
Tested by: Cao Minh Hiep

Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855

app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload.  This resulted in leaks in both
areas.

* app_voicemail now calls ast_delete_mwi_state_full when it frees
  a user structure and ast_delete_mwi_state_full in turn now calls
  the new stasis_topic_pool_delete_topic function to clear the topic
  from the pool.

Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8

The append_mailbox function wasn't calculating the correct length
to pass to ast_alloca and it wasn't handling the case where context
might be empty.

Found by the Address Sanitizer.

Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161

app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers.  It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled.  For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.

Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.

This paves the way for disabling the caching of stasis subscription
change messages.

Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.

ASTERISK-27121

Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee

This ensures the most up-to-date information is used for the next
call attempt.

ASTERISK-28032

Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce

Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8

I'm only seeing an error in 14+, so I assume it is due to different
compiler options:

app_queue.c: In function ‘handle_queue_add_member’:
app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11
    bytes into a region of size 3 [-Werror=format-overflow=]
     sprintf(num, "%d", state);
                   ^~
app_queue.c:10234:18: note: directive argument in the range
    [-2147483648, 99]
     sprintf(num, "%d", state);
                  ^~~~

Compiler: gcc version 8.0.1 20180414 (experimental)
    [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) 

Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10

When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.

The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.

ASTERISK-27973 #close
Reported-by: Valentin Safonov

Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c

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