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

Allow mapping pjproject log messages to the Asterisk TRACE
log level.  The defaults were also changes to log pjproject
levels 3,4 to DEBUG and 5,6 to TRACE.  Previously 3,4,5,6
all went to DEBUG.

ASTERISK-29582

Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d

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

The MIN, MAX, and ABS functions all support float
arguments, but currently return floats even if the
arguments are all integers and the response is
a whole number, in which case the user is likely
expecting an integer. This casts the float to an integer
before printing into the response buffer if possible.

ASTERISK-29495

Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488

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

Adds a function to scramble audio on a channel using
whole spectrum frequency inversion. This can be used
as a privacy enhancement with applications like
ChanSpy or other potentially sensitive audio.

ASTERISK-29542

Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e

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

Commit 305ce3de added -Wno-parentheses-equality to Makefile.rules,
turning the previous two warning suppressions from commit e9520dbe
redundant. Let us remove the latter.

Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac

Adds replacement modules to the moduleinfo for
chan_alsa and chan_sip.

ASTERISK-29601 #close

Change-Id: I7a4877b0d5c0c17e088e8fa8ebbfa9a195223cbc

ASTERISK-29602

Change-Id: I6f0af0a959409cdbc6b185b1604301bafc872a5a

ASTERISK-29600

Change-Id: I0ae1c6a2996da43217126f094de90761314dcf82

ASTERISK-29599

Change-Id: I75dc77162926fb17e7c6caf8f04e3aabd792fb0c

ASTERISK-29598

Change-Id: I8ef17023f55bf01f2e309b06f4778a8ca7252c91

ASTERISK-29597

Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1

ASTERISK-29596

Change-Id: Ibae9490c1b35cadbf7028d24610f745277c8535e

ASTERISK-29595

Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33

ASTERISK-29594

Change-Id: I79a9961cb5062fadbccb0ea93f087bdd32685316

ASTERISK-29593

Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99

ASTERISK-29592

Change-Id: Ic8eb6a2100ad5bc3b48338a6d0a6cfa70ecbc50f

ASTERISK-29591

Change-Id: I021d37b729631d40f84e35bb21e2893777be1858

ASTERISK-29590

Change-Id: I87cf0f536b77d222c8eda003376ac47fae86ed43

ASTERISK-29589

Change-Id: I8057eb2ca1ca4c3b27ed2fe04bea10e9cb551cdd

ASTERISK-29588

Change-Id: If846d40b37c5b646bcd7326111db280529a5971b

ASTERISK-29587

Change-Id: I038237bbb56b1161d7d5e20cda11ed32e13d3ca2

ASTERISK-29586

Change-Id: I1e0a4535135b00938b609fe0ccba9bbddbac93ad

ASTERISK-29585

Change-Id: I262930d0387d043f2a3345e8a977b314528059bf

ASTERISK-29584

Change-Id: I4bd3695d089121f810d692a82361d39d2f97ae39

ASTERISK-20339 #close

Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5

app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572

Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

Adds function to selectively drop specified frames
in the TX or RX direction on a channel, including
control frames.

ASTERISK-29478

Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec

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

This format did not specify a "write" handler, so when attempting to write
to it (ast_writestream) a crash would occur.

This patch adds a default handler that simply issues a "not supported"
warning, thus no longer crashing.

ASTERISK-29539

Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91

Previously, if CDR filters were used so that
not all CDR records used all sections defined
in cdr_adaptive_odbc.conf, then warnings will
always be emitted (if each CDR record is unique
to a particular section, n-1 warnings to be
specific).

This turns the offending warning log into
a verbose message like the other one, since
this behavior is intentional and not
indicative of anything wrong.

ASTERISK-29494

Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8

Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.

ASTERISK-29528

Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a

PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
It may be used to get all X- headers in case the actual set and names of headers unknown.

ASTERISK-29389

Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b

Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.

A flag has been introduced to allow meters to fallback to counters.


ASTERISK-29513

Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7

Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.

ASTERISK-29477

Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f

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

Asterisk first looks at the end of the URL to determine the file
extension of the returned audio, which in many cases will not work
because the URL may end with a query string or a URL fragment. If that
fails, Asterisk then looks at the Content-Type header and then finally
parses the URL to get the extension.

The order has been changed such that we look at the Content-Type
header first, followed by looking for the extension of the parsed
URL. We no longer look at the end of the URL, which was error prone.

ASTERISK-29527 #close

Change-Id: I1e3f83b339ef2b80661704717c23568536511032

Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c