There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.

ASTERISK-29632 #close

Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4

When compiled without extended srtp crypto suites also disable parsing
these from received SDP. This prevents using these, as some client
implementations are not stable.

ASTERISK-29625

Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a

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 conditions under which a failure to change topology
is expected so there's no need to print an ERROR message.

ASTERISK-29618
Reported by: Alexander

Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481

Fixed ARI external media handler to accept body parameters.

ASTERISK-29622

Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d

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

Adds an information element for ANI2 so that
Originating Line Information can be transmitted
over IAX2 channels.

ASTERISK-29605 #close

Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2

Currently pbx_ael does not check if a reload is currently pending
before proceeding with a reload. This can cause multiple threads to
operate at the same time on what should be mutex protected data. This
change adds protection to reloading to ensure only one ael reload is
executing at a time.

ASTERISK-29609 #close

Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35

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

Prevents reloads of app_queue from also resetting
queue statistics.

Also preserves individual queue agent statistics
if we're just reloading members.

ASTERISK-28701

Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1

ASTERISK-29616

Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900

Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83

This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.

In particular the behaviours that this impacts include:

 - FGC-CAMA did not work at all when using MF signaling. Modified the
   switch case block to send calls to the correct part of the
   signaling-handling state machine.

 - For FGC-CAMA operation, the delay between called number ST and
   second wink for ANI spill has been made configurable; previously
   all calls were made to wait for one full second.

 - After the ANI spill, previous behavior was to require a 'ST' tone
   to advance the call.  This has been changed to allow 'STP' 'ST2P'
   or 'ST3P' as well, for compatibility with ANI-D.

 - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.

 - For calls with an ANI failure, No. 1 Crossbar switches will send
   forward a single-digit failure code, with no calling number digits
   and no ST pulse to terminate the spill.  I've made the ANI timeout
   configurable so to reduce dead air time on calls with ANI fail.

 - ANI info digits configurable.  Modern digital switches will send 2
   digits, but ANI-B sends only a single info digit.  This caused the
   ANI reported by Asterisk to be misaligned.

 - Changed a confusing log message to be more informative.

ASTERISK-29518

Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256

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

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.
cdr_mysql was deprecated in 1.8, to be removed in 19.
app_mysql was deprecated in 1.8, to be removed in 19.
app_ices was deprecated in 16, to be removed in 19.
app_macro was deprecated in 16, to be removed in 21.
app_fax was deprecated in 16, to be removed in 19.
app_url was deprecated in 16, to be removed in 19.
app_image was deprecated in 16, to be removed in 19.
app_nbscat was deprecated in 16, to be removed in 19.
app_dahdiras was deprecated in 16, to be removed in 19.
cdr_syslog was deprecated in 16, to be removed in 19.
chan_oss was deprecated in 16, to be removed in 19.
chan_phone was deprecated in 16, to be removed in 19.
chan_sip was deprecated in 17, to be removed in 21.
chan_nbs was deprecated in 16, to be removed in 19.
chan_misdn was deprecated in 16, to be removed in 19.
chan_vpb was deprecated in 16, to be removed in 19.
res_config_sqlite was deprecated in 16, to be removed in 19.
res_monitor was deprecated in 16, to be removed in 21.
conf2ael was deprecated in 16, to be removed in 19.
muted was deprecated in 16, to be removed in 19.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29554
ASTERISK-29555
ASTERISK-29557
ASTERISK-29558
ASTERISK-29559
ASTERISK-29560
ASTERISK-29561
ASTERISK-29562
ASTERISK-29563
ASTERISK-29564
ASTERISK-29565
ASTERISK-29566
ASTERISK-29567
ASTERISK-29568
ASTERISK-29569
ASTERISK-29570
ASTERISK-29571
ASTERISK-29572
ASTERISK-29573
ASTERISK-29574

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

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

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

Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c

If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.

ASTERISK-29415 #close

Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.

This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.

ASTERISK-29392 #close

Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.

ASTERISK-29381

Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042