A regression was introduced in ASTERISK~29531 that caused 'say'
functions to fail with file lists that would previously have
succeeded. This caused affected channels to hang up where previously
they would have continued.

We now explicitly check for the empty string to restore the previous
behavior.

ASTERISK-29859 #close

Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5

Every config variable in the directories
section of asterisk.conf currently has a
counterpart built-in variable containing
the value of the config option, except
for the last one, astsbindir, which should
have an ASTSBINDIR variable.

However, the actual corresponding ASTSBINDIR
variable is missing in pbx_variables.c.

This adds the missing variable so that all
the config options have their corresponding
variable.

ASTERISK-29847 #close

Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc

Documentation for built-in special system and channel
vars is currently outdated, and updating is a manual
process since there is no XML documentation for these
anywhere.

This adds documentation for system vars to func_env
and for channel vars to func_channel so that they
appear along with the corresponding fields that would
be accessed using a function.

ASTERISK-29848 #close

Change-Id: I6997f925c4a45fffe71321861f5898a8b7182fa9

There are times when you need to troubleshoot issues with bundled
pjproject or add new features that need to be pushed upstream
but...

* The source directory created by extracting the pjproject tarball
  is not scanned for code changes so you have to keep forcing
  rebuilds.
* The source directory isn't a git repo so you can't easily create
  patches, do git bisects, etc.
* Accidentally doing a make distclean will ruin your day by wiping
  out the source directory, and your changes.
* etc.

This commit makes that easier.
See third-party/pjproject/README-hacking.md for the details.

ASTERISK-29824

Change-Id: Idb1251040affdab31d27cd272dda68676da9b268

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

Fixes 12pm noon incorrectly returning 0/a.m.
Also fixes a misspelling typo in the config.

ASTERISK-29695 #close

Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406

ASTERISK-29664 #close

Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5

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

Adds the macro DTMF_MATRIX_SIZE to replace
the magic number 4 sprinkled throughout
dsp.c.

ASTERISK-29815 #close

Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba

Adds an AMI event for a wink frame.

ASTERISK-29830 #close

Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56

Adds a command to the CLI to unload and then
load a module. This makes it easier to perform
these operations which are often done
subsequently to load a new version of a module.

"module reload" already refers to reloading of
configuration, so the name "refresh" is chosen
instead.

ASTERISK-29807 #close

Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c

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

The current TCP client connect code, blocks and does not handle EINTR
error case.

This patch makes the client socket non-blocking while connecting,
ensures a connect does not immediately fail due to EINTR "errors",
and adds a connect timeout option.

The original client start call sets the new timeout option to
"infinite", thus making sure old, orginal behavior is retained.

ASTERISK-29746 #close

Change-Id: I907571843a83e43c0742b95a64785f4411f02671

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

SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
and SayPhonetic all claim to allow DTMF interruption if the
SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
are failing to break out of a given 'say' application if DTMF actually
occurs.

ASTERISK-29816 #close

Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e

The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.

ASTERISK-28863
ASTERISK-29320

Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25

It's not safe to keep the channel locked while locking
the peer Local channel, as it can result in a deadlock.

This change unlocks it during this time but keeps the
bridge locked to ensure nothing changes about the bridge.

ASTERISK-29821

Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6

When test_timezone_watch runs very near a DST transition,
two time zones that would otherwise be expected to report the same
time can differ because of the DST transition.

Instead of having the test fail when this happens, report the
times, time zones, and dst flags.

ASTERISK-29722

Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a

Adding upstream patch for pull request...
https://github.com/pjsip/pjproject/pull/2920
---------------------------------------------------------------

sip_inv:  Additional multipart support (#2919)

sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart
message bodies in rdata correctly. In the case where early media is
involved though, the existing sdp has to be retrieved from the last
tdata sent in this transaction. This, however, always assumes that
the sdp sent is in a non-multipart body. While there's a function
to retrieve the sdp from multipart and non-multpart rdata bodies,
no similar function for tdata exists.  So...

* The existing pjsip_rdata_get_sdp_info2 was refactored to
  find the sdp in any body, multipart or non-multipart, and
  from either an rdata or tdata.  The new function is
  pjsip_get_sdp_info.  This new function detects whether the
  pjsip_msg->body->data is the text representation of the sdp
  from an rdata or an existing pjmedia_sdp_session object
  from a tdata, or whether pjsip_msg->body is a multipart
  body containing either of the two sdp formats.

* The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2
  functions are now wrappers that get the body and Content-Type
  header from the rdata and call pjsip_get_sdp_info.

* Two new wrappers named pjsip_tdata_get_sdp_info and
  pjsip_tdata_get_sdp_info2 have been created that get the body
  from the tdata and call pjsip_get_sdp_info.

* inv_offer_answer_test.c was updated to test multipart scenarios.

ASTERISK-29804

Change-Id: I483c7c3d413280c9e247a96ad581278347f9c71b

When OUTPUTDIR is set to another directory and the
--delete-results-after is set, the resulting txt files are
not deleted.

ASTERISK-29794 #close

Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286

The variable cp4 in a variable substitution function
can potentially be used without being initialized
currently. This causes Asterisk to no longer compile.

This initializes cp4 to NULL to make the compiler
happy.

ASTERISK-29803 #close

Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72

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

Previously, it was only possible to have one HTTP server in Asterisk.
With this patch it is now possible to have multiple HTTP servers
listening on different addresses.

Note, this behavior has only been made available through an API call
from within the TEST_FRAMEWORK. Specifically, this feature has been
added in order to allow unit test to create/start and stop servers,
if one has not been enabled through configuration.

Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5

Currently, Asterisk doesn't throw warnings if options
are passed into applications that don't accept them.
This can confuse users if they're unaware that they
are doing something wrong.

This adds an additional check to parse_options so that
a warning is thrown anytime an option is parsed that
doesn't exist in the parsing application, so that users
are notified of the invalid usage.

ASTERISK-29801 #close

Change-Id: Id029274a57135caca193c913307a63fd75e24679

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

The enum values for ast_strsep_flags includes
AST_STRSEP_STRIP. However, some comments reference
AST_SEP_STRIP, which doesn't exist. This fixes
these comments to use the correct value.

ASTERISK-29800 #close

Change-Id: If7bbd0c0e6226a211d25ddf9d1629347e2674943

Includes some minor updates to extensions.conf
and iax.conf. In particular, the demonstration
of macros in extensions.conf is removed, as
Macro is deprecated and will be removed soon.
These examples have been replaced with examples
demonstrating the usage of Gosub instead.

The older exten => ...,n syntax is also mostly
replaced with the same keyword to demonstrate the
newer, more concise way of defining extensions.

IAXTEL no longer exists, so this example is replaced
with something more generic.

Some documentation is also added to extensions.conf
and iax.conf to clarify some of the new expanded
encryption capabilities with IAX2.

ASTERISK-29758 #close

Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46

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

Currently MSet can only parse a maximum of 24 variables.
If more variables are provided to MSet, the 24th variable
will simply contain the remainder of the string and the
remaining variables thereafter will never get set.

This increases the number of variables that can be parsed
in one go from 24 to 99. Additionally, documentation is added
since this limitation is currently undocumented and is
confusing to users who encounter this limitation.

ASTERISK-29766 #close

Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16

Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
initiating an outgoing call will cause Asterisk to crash. This is
because a null field is accessed, resulting in an offset from null and
subsequent memory access violation.

Since RURI is not guaranteed to exist, we now check if the base
pointer is non-null before calculating an offset.

ASTERISK-29772

Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83

Adds the JSON_DECODE function for parsing JSON in the
dialplan. JSON parsing already exists in the Asterisk
core and is used for many different things. This
function exposes the basic parsing capability to
the user in the dialplan, for instance, in conjunction
with CURL for using API responses.

ASTERISK-29706 #close

Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c

Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd

Currently, variable substitution involving dialplan
extensions is quite clunky since it entails obtaining
the current dialplan location, backing it up, storing
the desired variables for substitution on the channel,
performing substitution, then restoring the original
location.

In addition to being clunky, things could also go wrong
if an async goto were to occur and change the dialplan
location during a substitution.

Fundamentally, there's no reason it needs to be done this
way, so new API is added to allow for directly passing in
the dialplan location for the purposes of variable
substitution so we don't need to mess with the channel
information anymore. Existing API is not changed.

ASTERISK-29745 #close

Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4

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

Otherwise, the value 'false' was not found in the enumerated set of
the XML DTD for the XML attribute 'required' in the XML element
'parameter'. Therefore, DTD validation of the runtime XML failed.

ASTERISK-29790

Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e

Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

ASTERISK-29791

Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725

In developer mode, use internal documentation as well.
This should produce no warnings. Fix yours!

In noisy mode, output all possible warnings of Doxygen.
This creates zillion of warnings. Double-check your current module!

Any warnings are in the file './doxygen.log'. Beside that, this change
avoids deprecated parameters because the configuration file for Doxygen
contains only those parameters which differ from the default. This
avoids the need to update the file on each run. Furthermore, it adds
AST_VECTOR to be expanded. Finally, the default name for that file is
Doxyfile. Therefore, let us use that!

ASTERISK-26991
ASTERISK-20259

Change-Id: I4129092a199d5e24c319a09cd088614b121015af