This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

The following modules needed tweaks for API changes.

addons/cdr_mysql.c
addons/chan_ooh323.c
apps/app_meetme.c

ASTERISK-28604

Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d

If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.

ASTERISK-22192

Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd

This reverts commit fd2e8d0d.

Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.

ASTERISK-28567 #close

Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41

Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a

ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.

This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().

ASTERISK-28527

Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1

There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:

 1. There is no rx and no tx audio, so return nothing.
 2. There is rx but no tx audio, so return rx.
 3. There is tx but no rx audio, so return tx.
 4. There is rx and tx audio, so mix them and return.

The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.

If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.

This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.

Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e

The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.

ASTERISK-28505 #close

Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c

You now select voicemail backends like normal dialplan applications, so
there is no longer a need for their own menuselect category.

Reported by snuff-work in #asterisk-dev

Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005

There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:

unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);

would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.

ASTERISK-28480

Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1

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