app_voicemail utilized the stasis cache when polling mailboxes for MWI. This
caused a memory leak (items were not being appropriately removed from the
cache), and subsequent slowdown in system processing. This patch removes the
stasis cache dependency, thus alleviating the memory leak. It does this by
utilizing the new MWI API that better manages state lifetime.

ASTERISK-28443
ASTERISK-27121

Change-Id: Ie89fedaca81ea1fd03d150d9d3a1ef3d53740e46

Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.

Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7

Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual
wait time instead of `framelength` is enough to fix timeout / TOOLONG issue.

ASTERISK-28419 #close

Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb

When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.

Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1

AttendedTransfer queues up attended transfer to the given extension.

This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer

[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
   same => n,Return()

[my_transfer]
include => default
;;;

This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
atxfer => *7

[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer

[custom_atxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,AttendedTransfer(${dest})
   same => n,Return()

[my_transfer]
include => default
;;;

Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b

BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.

This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.

features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer

[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
   same => n,Return()
;;;

This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:

features.conf
;;;
[featuremap]
blindxfer =>

[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;

extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer

[custom_blindxfer]
exten => s,1,
   same => n,Playback(pbx-transfer)
   same => n,Read(dest,dial,10,i,3,3)
   same => n,BlindTransfer(${dest},default)
   same => n,Return()
;;;

Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a

This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.

Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1

Various fixes for issues caught by gcc 9.  Mostly snprintf
trying to copy to a buffer potentially too small.

ASTERISK-28412

Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e

When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.

This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.

ASTERISK-28401

Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c

The total time logic will now be executed on calls which
do not pass any media.

ASTERISK-28143

Change-Id: I24726bd29d7e467fc721ca265363417234b22855

There a long history here:

In commit dd1e62c0 has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf

After that, the commit 4b50e3f1 fix the
sample file.

We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.

Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c

Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.

Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.

ASTERISK-28363

Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1

There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:

main/mwi.h
main/mwi.c

Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.

Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0

Because the per-mailbox options are the last thing on a line, don't look
for or stomp on any subsequent commas.

ASTERISK-27935 #close
Reported by: Sébastien Duthil

Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153

If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.

ASTERISK-20207 #close
Reported by: Steven Wheeler

Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf

* Always set member->lastpause when setting member->paused

* Fixed typo (using member->lastcall instead of member->lastpause) in
  'queue show' output.

* Use a constant 'now' in 'queue show' output for a better point-in-time
  view of time based stats.

ASTERISK-27541 #close
Reported by: César Benjamín García Martínez

Change-Id: Ib41ced90cfdb66f9bb1e7b263d0f6fc1ac6e18fa

It was a copy/paste of the QUEUE_MEMBER_COUNT function's synopsis.

ASTERISK-20986 #close
Reported by: Olivier Krief

Change-Id: If51ec481feb35824a4e78ab5600b197b819b10be

Fixes an intermittent segmentation fault which occured when accessing
nativeformats of a channel which entered into a queue.

ASTERISK-27964
Reported by: Francisco Seratti

Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e

Topic names now follow: <subsystem>:<functionality>[/<object>]

This ensures that they are all unique, and also provides better
insight in to what each topic is for.

Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.

Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.

Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.

ASTERISK-28335

Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d

ASTERISK-28328 #close

Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b

While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.

Also cleaned up an adjacent error message for consistency.

ASTERISK-28168 #close

Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202

This change add ability to set the wrapuptime per-member using the
AddQueueMember application.

The feature to set wrapuptime per member was include in the issue
ASTERISK-27483 for static member by configuration file and was not
added to set from AddQueueMember.

ASTERISK-28055 #close

Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf

When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.

This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.

This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.

ASTERISK-28277

Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6

You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes.  These can be used anywhere
the mailbox is specified.

Example:
[general]
aliasescontext = myaliases

[default]
1234 = yadayada

[myaliases]
4321@devices = 1234@default

Now you can use 4321@devices to refer to the 1234@default mailbox.

This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.

Change-Id: I395b9205c91523a334fe971be0d1de4522067b04

If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is
not updated correctly since urgent messages are in a different directory. The
fix is to update the channel variable when the path to the urgent message is
created.

ASTERISK-28225

Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca

When using the 'b' option to Queue with a queue that was not configured
for ring all a crash would occur as the wrong pointer would be used.

ASTERISK-28218

Change-Id: If1390f64e321047dff24fd2410c95dde74904980

The free_user function was automatically deleting the stasis mailbox
state but this only makes sense when the mailbox is actually
deleted, not just the structure freed.  This was causing issues
where leave_voicemail would publish the mwi message to stasis and
delete the state before the message could be processed by
res_pjsip_mwi.

* Removed the delete of state from free_user().

* Created a new free_user_final() function that both frees the data
  structure and deletes the state.  This function is only called
  during module load/unload where it's appropriate to delete the
  state.

ASTERISK-28215

Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd

Currently the file sound_only_person is not played when a marked
user (with announce_only_user=yes) joins an empty conference.

This patch fixes it.

ASTERISK-28201 #close

Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4

Revert commit 6409e7b1, and add
NULL checks for all app_queue event handling code.

Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844

ASTERISK-28125

Change-Id: I37334ea184ebb56e54471496b82937d4927815a0

This reverts commit 29115e23.

That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf.  This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.

ASTERISK-28151
Reported by: Ronald Raikes

Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15

* The bridging core no longer uses the stasis cache for bridge
  snapshots.  The latest bridge snapshot is now stored on the
  ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache
  is no longer used:
    ast_bridge_topic_cached()
    ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no
  longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the
  container of all bridges.  Users formerly calling
  ast_bridge_cache() can use the new function to iterate over
  bridges and retrieve the latest snapshot directly from the
  bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to
  ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve
  the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic
  not a cached one so you can't use stasis cache functions on it
  either.

* The ast_bridge_snapshot_type() stasis message now has the
  ast_bridge_snapshot_update structure as it's data.  It contains
  the last snapshot and the new one.

* cdr, cel, manager and ari have been updated to use the new
  arrangement.

Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369

When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.

This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.

The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.

Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.

ASTERISK-28119

Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423

Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.

As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()

The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.

The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.

The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.

ast_channel_snapshot_get_latest() still returns the latest snapshot.

The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.

ASTERISK-28102

Change-Id: I9334febff60a82d7c39703e49059fa3a68825786

Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.  Remove ao2_container_alloc macro.

Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088

This replaces the inline functions with macros.  This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.

Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b

This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.

This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.

There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.

ASTERISK-28103

Change-Id: I99bee23895baa0a117985d51683f7963b77aa190

The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.

This patch also blocks updates in the other direction before call is
answered.

ASTERISK-27980

Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01

Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel.  Only bridge_softmix has that
data so now it's set when the bridge topology is changed.

ASTERISK-28107

Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499

* Update the post-answer documentation and example.  The Dial example was
incorrect and misleading for the post-answer subroutine useage.

* Fix note and warning paragraphs in option descriptions.  They don't show
up in the wiki.

Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14

Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push.  This will help
ensure we perform the necessary cleanup upon failure.

Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d