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

Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.

This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.

ASTERISK-29861 #close

Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d

Refactors temp file utility function into file.c.

ASTERISK-29809 #close

Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b

Currently, if VoiceMailMain is called with a mailbox, if that
mailbox doesn't exist, then the application silently falls back
to prompting the user for the mailbox, as if no arguments were
provided.

However, if a specific mailbox is requested and it doesn't exist,
then no warning at all is emitted.

This fixes this behavior to now warn if a specifically
requested mailbox could not be accessed, before falling back to
prompting the user for the correct mailbox.

ASTERISK-29920 #close

Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4

If Subscription refresh occurred between when the batched notification
was scheduled and the serialized notification was to be sent,
then new schedule notification task would never be added.

There are 2 threads:

thread #1. ast_sip_subscription_notify is called,
if notification_batch_interval then call schedule_notification.
1.1. The schedule_notification checks notify_sched_id > -1
not true, then
send_scheduled_notify = 1
notify_sched_id =
  ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
1.2. The sched_cb pushes task serialized_send_notify to serializer
and returns 0 which means no reschedule.
1.3. The serialized_send_notify checks send_scheduled_notify if it's false
the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.

thread #2. pubsub_on_rx_refresh is called
2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
which calls send_notify
2.3. The send_notify set send_scheduled_notify = 0;

The serialized_send_notify should always unset notify_sched_id.

ASTERISK-29904 #close

Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875

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 a simple sanity check for key names when users are
writing data to AstDB. This captures four cases indicating
malformed keynames that generally result in bad data going
into the DB that the user didn't intend: an empty key name,
a key name beginning or ending with a slash, and a key name
containing two slashes in a row. Generally, this is the
result of a variable being used in the key name being empty.

If a malformed key name is detected, a warning is emitted
to indicate the bug in the dialplan.

ASTERISK-29925 #close

Change-Id: Ifc08a9fe532a519b1b80caca1aafed7611d573bf

Adds two pieces of information to the core show settings command
which are useful in the context of getting backtraces.

The first is to display whether or not Asterisk would generate
a core dump if it were to crash.

The second is to show the current running directory of Asterisk.

ASTERISK-29866 #close

Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83

The configObject tag contains a default attribute which
allows the default value to be specified, if applicable.
This allows for the default value to show up specially on
the wiki in a way that is clear to users.

There are a couple places in the tree where default values
are included in the description as opposed to as attributes,
which means these can't be parsed specially for the wiki.
These are changed to use the attribute instead of being
included in the text description.

ASTERISK-29898 #close

Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755

mpg123 doesn't support HTTPS, but the MP3Player application
doesn't document this or warn the user about this. HTTPS
streams have become more common nowadays and users could
reasonably try to play them without being aware they should
use the HTTP stream instead.

This adds documentation to note this limitation. It also
throws a warning if users try to use the HTTPS stream to
tell them to use the HTTP stream instead.

ASTERISK-29900 #close

Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827

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

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

When tps_shutdown is called as part of the cleanup process there is a
chance that one of the taskprocessors that references the
tps_singletons object is still running.  The change is to allow for
tps_shutdown to check tps_singleton's container count and give the
running taskprocessors a chance to finish.  If after
AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still
container references we shutdown anyway as this is most likely a bug
due to a taskprocessor not being unreferenced.

ASTERISK-29365

Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1

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

This code was needlessly complex and would fail to properly delimit
the response message if LOW_MEMORY was defined.

Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e

added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO
which fixes issue with module crashes on load "FRACK!, Failed assertion"

ASTERISK-29871

Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096

Change-Id: I886936b808def5540d40071321e72f6bfa19063a

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

In POSIX sh (which we indicate in the shebang), there is no ==
operator.

Change-Id: Ic03d38214d14cdf329b0ba272279a815bb532965

Added the following APIs:
pjsip_multipart_find_part_by_header()
pjsip_multipart_find_part_by_header_str()
pjsip_multipart_find_part_by_cid_str()
pjsip_multipart_find_part_by_cid_uri()

Change-Id: I6aee3dcf59eb171f93aae0f0564ff907262ef40d

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

ASTERISK-29832

Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841

If you have a development branch for a major project that
will receive gerrit reviews it'll probably be named something
like "development/16/newproject".  That will necessitate setting
"defaultbranch=development/16/newproject" in .gitreview.  The
make_version script uses that variable to construct the asterisk
version however, which results in versions like
"GIT-development/16/newproject-ee582a8c7b" which is probably not
what you want.  Worse, since the download_externals script uses
make_version to construct the URL to download the binary codecs
or DPMA.  Since it's expecting a simple numeric version, the
downloads will fail.

To get this to work, a new variable "basebranch" has been added
to .gitreview and make_version has been updated to use that instead
of defaultversion:

.gitreview:
defaultbranch=development/16/myproject
basebranch=16

Now git-review will send the reviews to the proper branch
(development/16/myproject) but the version will still be
constructed using the simple branch number (16).

If "basebranch" is missing from .gitreview, make_version will
fall back to using "defaultbranch".

Change-Id: I2941a3b21e668febeb6cfbc1a7bb51a67726fcc4

In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess
that hasn't been initialized, it'll assert and abort.  If
digest_create_request_with_auth() fails to find the proper
auth object however, it jumps to its cleanup which does exactly
that.  So now we no longer attempt to call pjsip_auth_clt_deinit()
if we never actually initialized it.

ASTERISK-29888

Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62

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

Fixes some minor logic issues with the module:

Previously, the OPT_END_FILTER flag was getting
tested before options were parsed, so it could
never evaluate to true (wrong ordering).

Additionally, the initially parsed timeout (float)
needs to be compared with 0, not the result int
which is set afterwards (wrong variable).

ASTERISK-29857 #close

Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c

In order to get around the issue of certain frames
having names that could overlap, func_frame_drop
surrounds names with commas for the purposes of
comparison.

The buffer is allocated and printed to properly,
but the original buffer is used for comparison.
In most cases, this wouldn't have had any effect,
but that was not the intention behind the buffer.
This updates the code to reference the modified
buffer instead.

ASTERISK-29854 #close

Change-Id: I430b52e14e712d0e62a23aa3b5644fe958b684a7

When generating dtmfs, asterisk can incorrectly think packet loss
occured during the dtmf generation, resulting in a jump in sequence
numbers when forwarding voice frames resumes.  This patch forces
asterisk to re-learn the expected sequence number after each DTMF
to avoid this

ASTERISK-29869 #close

Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853

Previously there was no way to specify a connection timeout when
attempting to connect a websocket client to a server. This patch
makes it possible to now do such.

Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d

autoconfigh.h.in was missed in the original review for this
issue. Additionally it looks like I have newer pkg-config autoconf
macros on my development machine.

ASTERISK-29817

Change-Id: I3c85a4de82c5d7d6e0e23dad4c33bb650a86a57b

sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an
executing call-back. This is done by adding a new variable 'rescheduled'
to the struct sched which is set in ast_sched_runq and checked in
ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for
now deprecated ast_sched_del which returns a new possible value -2
if called on an executing call-back with rescheduled set. ast_sched_del
is modified to call ast_sched_del_nonrunning to maintain existing code.
AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it
will not throw a warning or invoke refcall.
test_sched: Add a new unit test sched_test_freebird that will check the
reference count in the resolved scenario.

ASTERISK-29698

Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d

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

ASTERISK-28890 #close

Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1

Fix the sed(1) invocation used to process git-svn-id not to use "\s"
that is a GNU-ism and is not supported by NetBSD sed.  As a result,
this call did not work properly and make_version did output the full
git-svn-id line rather than the revision.

ASTERISK-29852

Change-Id: Ie4b406e2748920643446851a0a252a4ca7245772

Implement the ast_get_tid() function for NetBSD system.  NetBSD supports
getting the TID via _lwp_self().

ASTERISK-29850

Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c

Enable the Linux rdtsc implementation on NetBSD as well.  The assembly
works correctly there.

ASTERISK-29851

Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37

Fix the configure script not to detect the presence of gethostbyname_r()
on NetBSD incorrectly.  NetBSD includes it as an internal libc symbol
that is not exposed in system headers and that is incompatible with
other implementations.  In order to avoid misdetecting it, perform
the symbol check only if the declaration is found in the public header
first.

ASTERISK-29817

Change-Id: Iafa359b09908251bcd299ff54be003ea129b9eda

Remove the HMAC declarations from the includes.  They are
not implemented nor used anywhere, and their presence breaks the build
on NetBSD that delivers an incompatible hmac() function in <stdlib.h>.

ASTERISK-29818

Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7

Fixes CNG description from "noice" to "noise".

ASTERISK-29855 #close

Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d

The code currently checks to see if an RFC3389
warning flag is set, except if it is, it merely
sets the flag again, the logic of which doesn't
make any sense.

This adjusts the if comparison to check if the
flag has NOT been set, and if so, emit a notice
log event and set the flag so that future frames
do not cause an event to be logged.

ASTERISK-29856 #close

Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1

Reverted recent change that set '--with-external-srtp' instead
of '--without-external-srtp'.  Since Asterisk handles all SRTP,
we don't need it enabled in pjproject at all.

ASTERISK-29867

Change-Id: I2ce1bdd30abd21c062eac8f8fefe9b898787b801