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

Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.

This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.

Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.

ASTERISK-29531

Change-Id: If9718c89353b8e153d84add3cc4637b79585db19

dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.

Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.

ASTERISK-29546

Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26

ncurses 6.1 introduced an extended number format for terminfo files
which the terminfo parsing in Asterisk is not able to parse. This
results in some TERM values that do support color (screen-256color on
Ubuntu 20.04 for example) to not get a color console.

ASTERISK-29630 #close

Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3

IPv6 nameserver addresses are stored in different part of the
__res_state structure, so look there if we appear to have support for
it.

ASTERISK-28004 #close

Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d

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

This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.

The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.

ASTERISK-29508 #close

Change-Id: I7955a046293f913ba121bbd82153b04439e3465f

The attended transfer feature will emit a warning if the user
cancels the transfer or the attended transfer doesn't complete
for any reason. Changes the warning to a verbose message,
since nothing is actually wrong here.

ASTERISK-29612 #close

Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d

When playing a remote sound file, which is not in cache, first we need
to download it with ast_bucket_file_retrieve.

This can take a while if the remote host is slow. The current CURL
timeout is 180secs, so in extreme situations, it can take 3 minutes to
return.

Because ast_media_cache_retrieve has a lock on all function, while we
are waiting for the delayed download, Asterisk is not able to play any
more files, even the files already cached locally.

ASTERISK-29544 #close

Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408

With Asterisk 1.6.0, in the main parser for the configuration file
extensions.conf, the separator was changed from vertical bar to comma.
However, the first separator was not changed in aelparse; it still had
to be a vertical bar, and no comma was allowed.

Additionally, this change allows the vertical bar for the first and
last parameter again, even in the main parser, because the vertical bar
was still accepted for the other parameters.

ASTERISK-29540

Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81

If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
This makes the audio stream not-playable at the receiver side.
Linphone isn't able to play such an audio - lots of disruptions are heard.
Also I had complains of bad audio from users which use other types of phones.

After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).

Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).

However, this flag is never set in Asterisk-12 and newer.
Previously it has been set (see Asterisk-11).

ASTERISK-29526 #close

Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d

Use cURL's URL parsing API, falling back to the urlparser library, to
parse playback URLs in order to find their file extensions.

For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.

ASTERISK-27871 #close

Change-Id: I16d0682f6d794be96539261b3e48f237909139cb

This appears to just have been a copy/paste error from 6258bbe7. Fix
suggested by Ross Beer in ASTERISK~29166.

Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37

Without this message, it is not obvious that the reason is STUN timeout.

ASTERISK-29507 #close

Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74

If the system time has stepped backwards because of a time
adjustment between the time a frame is timestamped and the
time we check the timestamps in abstract_jb:hook_event_cb(),
we get a negative interval, but we don't check for that there.
abstract_jb:hook_event_cb() then calls
fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
and the first thing that does is assert(interval >= 0).

There are several issues with this...

 * abstract_jb:hook_event_cb() saves the interval in a variable
   named "now" which is confusing in itself.

 * "now" is defined as an unsigned int which converts the negative
   value returned from ast_tvdiff_ms() to a large positive value.

 * fixed_jb_get()'s parameter is defined as a signed int so the
   interval gets converted back to a negative value.

 * fixed_jb_get()'s assert is NOT an ast_assert but a direct define
   that points to the system assert() so it triggers even in
   production mode.

So...

 * hook_event_cb()'s "now" was renamed to "relative_frame_start" and
   changed to an int64_t.
 * hook_event_cb() now checks for a negative value right after
   retrieving both the current and framedata timestamps and just
   returns the frame if the difference is negative.
 * fixed_jb_get()'s local define of ASSERT() was changed to call
   ast_assert() instead of the system assert().

ASTERISK-29480
Reported by: Dan Cropp

Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9

When using the Busy() and Congestion() applications the
function ast_safe_sleep is used by wait_for_hangup to safely
wait on the channel. This function may send silence if Asterisk
is configured to do so using the transmit_silence option.

In a scenario where an answered channel dials a Local channel
either directly or through call forwarding and the Busy()
or Congestion() dialplan applications were executed with the
transmit_silence option enabled the busy or congestion
tone would not be heard.

This is because inband generation of tones (such as busy
and congestion) is stopped when other audio is sent to
the channel they are being played to. In the given
scenario the transmit_silence option would result in
silence being sent to the channel, thus stopping the
inband generation.

This change adds a variant of ast_safe_sleep which can be
used when silence should not be played to the channel. The
wait_for_hangup function has been updated to use this
resulting in the tones being generated as expected.

ASTERISK-29485

Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133

Previously, SayNumber always emitted a warning if the caller hung up
during execution. Usually this isn't correct, so check if the channel
hung up and, if so, don't emit a warning.

ASTERISK-29475

Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594

Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575

Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.

This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.

ASTERISK-29380

Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81

AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.

ASTERISK-29372

Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc

STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.

The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c

Enhancements:

 * The MessageSend dialplan application now takes an optional
   third argument that can set the message's "To" field on
   outgoing messages.  It's an alternative to using the
   MESSAGE(to) dialplan function.

   NOTE: No channel driver currently implements this field.  A
   follow-on commit for res_pjsip_messaging will implement it for
   the chan_pjsip channel driver.

 * To prevent confusion with the first argument, currently named
   "to", it's been renamed to "destination". Its function,
   creating the request URI, hasn't changed.

 * The documentation for MessageSend was updated to be
   more clear about the parameters and how they interact
   the MESSAGE() dialplan function.

 * With the rename of MessageSend's first parameter, and the fact
   that message.c references <info> elements in chan_sip.c,
   res_pjsip_messaging.c and res_xmpp, they each needed
   documentation updates to use MessageDestinationInfo instead of
   MessageToInfo.

 * appdocsxml.dtd was updated to include a missing element
   declaration for "dataType".  This was showing up as an error
   in Eclipse's dtd editor.

 * Despite the changes in this commit, there should be
   no impact to current users of MessageSend.

Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a

Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6

When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.

This change causes such streams to be skipped.

ASTERISK-29407

Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6

Up/down sampling changes the number of samples produced by a translation.
This must be taken into account when checking the codec's buffer size.

ASTERISK-29328

Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055

There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL.  The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff.  So...

bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.

As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.

ASTERISK-29379
Reported-by: Vyrva Igor
Reported-by: Ross Beer

Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce

Only use an XPath query once per module, then just navigate the DOM for
everything else.

Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92

Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.

ASTERISK-29339

Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a

Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.

Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392

The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.

Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
  main/logger.c: static char dateformat[256] = "%b %e %T"

This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages

Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko

When using the ast_unreal_lock_all function no channel
locks can be held before calling it.

This change unlocks the channel that indicate was
called on before doing so and then relocks it afterwards.

ASTERISK-29035

Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9

There exists an inconsistency with framehook usage
such that it is only on reads that the frame should
be freed, not on writes as well.

ASTERISK-29071

Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472

Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.

This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.

ASTERISK-29321

Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1

ASTERISK-29326

Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a

A frame suppression API exists as part of channels
which allows audio frames to or from a channel to
be dropped. The MuteAudio AMI action uses this
API to perform its job.

This API uses a framehook to intercept flowing
audio and drop it when appropriate. It is the
responsibility of the framehook to free the
frame it is given if it changes the frame. The
suppression API failed to do this resulting in
a leak of audio frames.

This change adds the freeing of these frames.

ASTERISK-29071

Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b

Implemented the english way of saying the year in ast_say_date_with_format_nl.
Currently the numbers are spoken correctly until 2020 and stopped working
this year.

ASTERISK-29297 #close
Reported-by: Jacek Konieczny

Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4

Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.

This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.

Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34

ASTERISK-29244

Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5

After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.

ASTERISK-29035 #close

Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb