This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.

ASTERISK-29909 #close

Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec

This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.

ASTERISK-29906 #close

Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3

The disabledevents setting has been added to the general section
in manager.conf, which allows users to specify events that
should be globally disabled and not sent to any AMI listeners.

This allows for processing of these AMI events to end sooner and,
for frequent AMI events such as Newexten which users may not have
any need for, allows them to not be processed. Additionally, it also
cleans up core debug as previously when debug was 3 or higher,
the debug was constantly spammed by "Analyzing AMI event" messages
along with a complete dump of the event contents (often for Newexten).

ASTERISK-29853 #close

Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205

Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.

ASTERISK-29840 #close

Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307

Adds the since tag to the documentation DTD so
that individual applications, functions, etc.
can now specify when they were added to Asterisk.

This tag is added at the individual application,
function, etc. level as opposed to at the module
level because modules can expand over time as new
functionality is added, and granularity only
to the module level would generally not be useful.

This enables the ability to more easily determine
when new functionality was added to Asterisk, down
to minor version as opposed to just by major version.
This makes it easier for users to write more portable
dialplan if desired to not use functionality that may
not be widely available yet.

ASTERISK-29896 #close

Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5

Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.

This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.

ASTERISK-29891 #close

Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681

Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.

Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.

This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).

ASTERISK-29877 #close

Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b

There are a lot of Queue AMI actions and Queue applications
which do not load queue and queue members from Realtime.

AMI actions
QueuePause - if queue not in memory - response "Interface not found".
QueueStatus/QueueSummary - if queue not in memory - empty response.

Applications:
PauseQueueMember - if queue not in memory
	Attempt to pause interface %s, not found
UnpauseQueueMember - if queue not in memory
	Attempt to unpause interface xxxxx, not found

This patch adds a new function load_realtime_queues
which loads queue and queue members for desired queue
or all queues and all members if param 'queuename' is NULL or empty.
Calls the function load_realtime_queues when needed.

Also this patch fixes leak of ast_config in function set_member_value.

Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
already paused/unpaused member.
The function ast_update_realtime returns 0 when no record modified.
So 0 is not an error to warn about.

ASTERISK-29873 #close
ASTERISK-18416 #close
ASTERISK-27597 #close

Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c

The XML Manager Event Interface (amxml) now generates attribute names
that are compliant with the XML 1.1 specification. Previously, an
attribute name that started with a digit would be rendered as-is, even
though attribute names must not begin with a digit. We now prefix
attribute names that start with a digit with an underscore ('_') to
prevent XML validation failures.

This is not backwards compatible but my assumption is that compliant
XML parsers would already have been complaining about this.

ASTERISK-29886 #close

Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523

resolve issue with pickup on device that uses "183" and not "180"

ASTERISK-29832

Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841

Adds a new option, defaultenabled, to the CDR core to
control whether or not CDR is enabled on a newly created
channel. This allows CDR to be disabled by default on
new channels and require the user to explicitly enable
CDR if desired. Existing behavior remains unchanged.

ASTERISK-29808 #close

Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1

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

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

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 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

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 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

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

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

Newer versions of spandsp did refactoring of code to add new features
like color FAXing. This refactoring broke backwards compatibility.
Add support for the new version while retaining support for 0.0.6.

ASTERISK-29729 #close

Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1

Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.

ASTERISK-29720 #close

Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90

Correct typos of the following word families:

transparent
roughly

ASTERISK-29714

Change-Id: I2b90c68dfde4aa3f0d58f64f8187465336acb1b3

Historically, the dial syntax for IAX2 has held that
an outkey (used only for RSA authenticated calls)
and a secret (used only for plain text and MD5 authenticated
calls, historically) were mutually exclusive, and thus
the same position in the dial string was used for both
values.

Now that encryption is possible with RSA authentication,
this poses a limitation, since encryption requires a
secret and RSA authentication requires an outkey. Thus,
the dial syntax is extended so that both a secret and
an outkey can be specified.

The new extended syntax is backwards compatible with the
old syntax. However, a secret can now be specified after
the outkey, or the outkey can be specified after the secret.
This makes it possible to spawn an encrypted RSA authenticated
call without a corresponding peer being predefined in iax.conf.

ASTERISK-29707 #close

Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e

The search for a running asterisk when --running is used
has been greatly simplified and in the event it doesn't
work, you can now specify a pid to use on the command
line with --pid.

The search for asterisk modules when --tarball-coredumps
is used has been enhanced to have a better chance of finding
them and in the event it doesn't work, you can now specify
--libdir on the command line to indicate the library directory
where they were installed.

The DATEFORMAT variable was renamed to DATEOPTS and is now
passed to the 'date' utility rather than running DATEFORMAT
as a command.

The coredump and output files are now renamed with DATEOPTS.
This can be disabled by specifying --no-rename.

Several confusing and conflicting options were removed:
--append-coredumps
--conffile
--no-default-search
--tarball-uniqueid

The script was re-structured to make it easier for follow.

Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59

The stir_shaken configuration option now has 4 different choices to pick
from: off, attest, verify, and on. Off and on behave the same way they
do now. Attest will only perform attestation on the endpoint, and verify
will only perform verification on the endpoint.

Certain responses are required to be sent based on certain conditions
for STIR/SHAKEN. For example, if we get a Date header that is outside of
the time range that is considered valid, a 403 Stale Date response
should be sent. This and several other responses have been added.

Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7

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

Adds support for encryption to RSA-authenticated
calls. Also prevents crashes if an RSA IAX2 call
is initiated to a switch requiring encryption
but no secret is provided.

ASTERISK-20219

Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40

In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.

ASTERISK-29402

Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557

The behavior of max_contacts and remove_existing are connected.  If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact.  Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.

This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing.  If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.

ASTERISK-29525

Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784

Allows multiple mailboxes to be specified for VMCOUNT
instead of just one.

ASTERISK-29661 #close

Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813

The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.

ASTERISK-29663 #close

Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c