if holdtime is (0 min, 0 sec) there is no hold time announcements
we should then also not playing queue-thankyou

ASTERISK-29831

Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c

gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.

ASTERISK-29819 #close

Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7

adding support for playing the correct en/et for nordic languages
by adding 'n' for neuter gender in the relevant ast_say_number

ASTERISK-29827

Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1

Currently, the MP3Player application doesn't
emit a warning if attempting to play a stream
which no longer exists. This can be a common
scenario as many mp3 streams are valid at some
point but can disappear at any time.

Now a warning is thrown if attempting to play
a nonexistent MP3 stream, instead of silently
exiting.

ASTERISK-29829 #close

Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535

Adds missing documentation for some channel,
bridge, and queue events.

ASTERISK-24427
ASTERISK-29515

Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64

Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.

ASTERISK-29802 #close

Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad

A previous patch for ASTERISK_29578 caused a 'leak' of
extension state information across queues, causing the
state of the first member of unrelated queues to be
updated in addition to the correct member. Which queues
and members depended on the order of queues in the
iterator.

Additionally, it is possible to use the same 'hint:' on
multiple queue members, so the update cannot break out
of the update loop early when a match is found.

ASTERISK-29806 #close

Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad

added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial

ASTERISK-29795

Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6

added support for playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code

ASTERISK-29797

Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2

Adds a ReceiveText application that can be used in
conjunction with SendText. Currently, there is no
way in Asterisk to receive text in the dialplan
(or anywhere else, really). This allows for Asterisk
to be the recipient of text instead of just the sender.

ASTERISK-29759 #close

Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d

Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.

ASTERISK-29496-mf #do-not-close

Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4

Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING

ASTERISK-28053
Reported by: roadkill

Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89

ASTERISK-29779

Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.

This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.

ASTERISK-29777 #close

Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b

Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.

ASTERISK-29715 #close

Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee

ASTERISK-29740

Change-Id: Icb6fbcfea0a5f1c82caa5001902b6a786adbf307

Fixes a deadlock in app_morsecode caused by locking
the channel twice when reading variables from the
channel. The duplicate lock is simply removed.

ASTERISK-29744 #close

Change-Id: I204000701f123361d7f85e0498fedc90243c75e4

Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.

This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.

Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.

ASTERISK-29705 #close

Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086

Correct typos of the following word families:

simultaneously
administrator
directforward
attachfmt
dailplan
automatically
applicable
nouns
explicit
outside
sponsored
attachment
audio
spied
doesn't
counting
encoded
implements
recursively
emailaddress
arguments
queuerules
members
priority
output
advanced
silencethreshold
brazilian
debugging
argument
meadmin
formatting
integrated
sneakiness

ASTERISK-29714

Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092

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