If users are able to press # for options while leaving
a message and then press 3 to rerecord the message, if
the caller hangs up during the rerecord prompt but before
Asterisk starts recording a message, then an "empty"
voicemail gets processed whereby an email gets sent out
notifying the user of a 0:00 duration message. The file
doesn't actually exist, so playback will fail since there
was no message to begin with.

This adds a check after the streaming of the rerecord
announcement to see if the caller has hung up. If so,
we bail out early so that we can clean up properly.

ASTERISK-29391 #close

Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31

Add a time_t logintime to storage a time when a member is added into a
queue.

Also, includes show this time (in seconds) using a 'queue show' command
and the field LoginTime for response for AMI events.

ASTERISK-18069 #close

Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190

I am adding a mix option that will play by filename and say.conf unlike
say option that will only play with say.conf. It
will look on the format of the name, if it is like say it play with
say.conf if not it will play the file name.

ASTERISK-29662

Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a

If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.

ASTERISK-29673 #close

Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac

Previously, if custom hints were used with the hint:
format in app_queue, when device state changes occured,
app_queue would only do a literal string comparison of
the context used for the hint in app_queue and the context
of the hint which just changed state. This caused hints
to not update and become stale if the context associated
with the agent included the context which actually changes
state, essentially completely breaking device state for
any such agents defined in this manner.

This fix adds an additional check to ensure that included
contexts are also compared against the context which changed
state, so that the behavior is correct no matter whether the
context is specified to app_queue directly or indirectly.

ASTERISK-29578 #close

Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78

Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.

ASTERISK-29529

Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702

No functional changes.

Change-Id: I46514152c0af67f395526374aaa847ccd6a85378

In new mpg123 versions (since 1.26) the default output is 32 bits
Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
It will work wit new and old versions of mpg123.
Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!

ASTERISK-29635 #close

Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816

Adds a SendMF application and PlayMF manager
event to send arbitrary R1 MF tones on the
current or specified channel.

ASTERISK-29496

Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4

Previously, the error emitted when app_stack tries
to branch to a dialplan location that doesn't exist
has included only the information about the attempted
branch in the error log. This adds the current location
as well so users can see where the branch failed in
the logs.

ASTERISK-29626

Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce

There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.

ASTERISK-29632 #close

Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4

There are 3 separate changes here but they are all closely related:

* Only try to set matchfield attributes on 'field' nodes

* We need to adjust how we treat the category pointer based on the
  value of the category_match, to avoid memory corruption. We now
  generate a regex-like string when match types other than
  ACO_WHITELIST and ACO_BLACKLIST are used.

* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
  ACO_BLACKLIST_EXACT since we only have one category we need to
  ignore, not two.

ASTERISK-29614 #close

Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e

Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.

ASTERISK-18454 #close

Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b

Prevents reloads of app_queue from also resetting
queue statistics.

Also preserves individual queue agent statistics
if we're just reloading members.

ASTERISK-28701

Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1

The Milliwatt application uses incorrect tone timings
that cause it to play the 1004 Hz tone constantly.

This adds an option to enable the correct timing
behavior, so that the Milliwatt application can
be used for milliwatt test lines. The default behavior
remains unchanged for compatability reasons, even
though it is incorrect.

ASTERISK-29575 #close

Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c

Previously, the Morsecode application only supported international
Morse code. This adds support for American Morse code and adds an
option to configure the frequency used in off intervals.

Additionally, the application checks for hangup between tones
to prevent application execution from continuing after hangup.

ASTERISK-29541

Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4

A list of codecs to use for dialplan-originated calls can
now be specified in Originate, similar to the ability
in call files and the manager action.

Additionally, we now default to just using the slin codec
for originated calls, rather than all the slin* codecs up
through slin192, which has been known to cause issues
and inconsistencies from AMI and call file behavior.

ASTERISK-29543

Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883

ASTERISK-29591

Change-Id: I021d37b729631d40f84e35bb21e2893777be1858

ASTERISK-29590

Change-Id: I87cf0f536b77d222c8eda003376ac47fae86ed43

ASTERISK-29589

Change-Id: I8057eb2ca1ca4c3b27ed2fe04bea10e9cb551cdd

ASTERISK-29588

Change-Id: If846d40b37c5b646bcd7326111db280529a5971b

ASTERISK-29587

Change-Id: I038237bbb56b1161d7d5e20cda11ed32e13d3ca2

ASTERISK-29586

Change-Id: I1e0a4535135b00938b609fe0ccba9bbddbac93ad

app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572

Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.

ASTERISK-29528

Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a

Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.

ASTERISK-29477

Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f

Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c

Adds an application to reload modules
from within the dialplan.

ASTERISK-29454

Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774

While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.

ASTERISK-29444

Change-Id: I08adf2824b8bc63405778cf355963b5005612f41

Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.

ASTERISK-29442

Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e

Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.

ASTERISK-29450

Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66

Adds a new ConfKick() application, which may
be used to kick a specific channel, all channels,
or all non-admin channels from a specified
conference bridge, similar to existing CLI and
AMI commands.

ASTERISK-29446

Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b

A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.

ASTERISK-29440

Change-Id: I26642729d0345f178c7b8045506605c8402de54b

Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.

To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.

ASTERISK-29349

Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280

Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.

ASTERISK-26614 #close

Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217

ASTERISK-27542 #close

Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a

If a queue member was updated with the same status multiple
times each time a QueueMemberStatus event would be sent
which would be a duplicate of the previous.

This change makes it so that the QueueMemberStatus event is
only sent if the status actually changes.

ASTERISK-29355

Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116

This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.

ASTERISK-29335

Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90

Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.

ASTERISK-29336

Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35

ASTERISK-29329 #close

Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601