Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.

Some notes:

* Because of the version of flex we are using to generate the lexers
  (2.5.35) some post-processing in the Makefile is still required.

* The generated lexers do not contain the problematic C99 check that
  was being replaced by the call to sed in the respective Makefiles so
  it was removed.

* Since these files are generated, they will include trailing
  whitespace in some places. This does not need to be corrected.

Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e
(cherry picked from commit 34ff836db5679cb1dc6f9d5ba8809d1a437ec0e3)

Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.

If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed.  If the interval is set to an
invalid value, it will default to 15 seconds.

A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started.  This is so we can test that the touch
variable starts the hook as expected.

ASTERISK-30446

Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661
(cherry picked from commit ffe346b2de8d175ba60e0860546c32c25cb88d9f)

While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.

This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute.  This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.

As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.

ASTERISK-30464

Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98
(cherry picked from commit fa635a872ea410d656d1f912a49bae66e95f1ae9)

Adds '.slin' to existing supported file extensions:
.sln and .raw

ASTERISK-30465

Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496
(cherry picked from commit 8d2ffc8aa54315b937712bc725fd813a37e73158)

Sending the "RECORD FILE" command without the optional
`offset_samples` argument can result in two beeps playing on the
channel.

This bug has been present since Asterisk 0.3.0 (2003-02-06).

ASTERISK-30457 #close

Change-Id: I95e88aa59378784d7f0eb648843f090e6723b787
(cherry picked from commit a3ec3efa02f1b40336f9192af0cb1e11f0b8bee8)

Fix issue with returning empty instead of dumping
the JSON string when recursing.

Also adds a unit test to capture this fix.

ASTERISK-30441 #close

Change-Id: If0bde9f3fe84f7af485e0838205cc21e0f752a85
(cherry picked from commit f9fd76677f20f9db05c26ac55cc064394feb8650)

Adds an AMI action to send a flash event
on a channel.

ASTERISK-30440 #close

Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9
(cherry picked from commit 3556ca239aa4b48eb31df4e19f54571d1ab3bd14)

DTMF frames are not handled in app_dial when sent towards the
caller. This means that if DTMF is sent to the calling party
and the call has not yet been answered, the DTMF is not audible.
This is now fixed by relaying DTMF frames if only a single
destination is being dialed.

ASTERISK-29516 #close

Change-Id: Iafd7430ac2915126d42dc48f0b73b262452ee027
(cherry picked from commit 090ec448cffc4b73e7840347aa148b5fc9181e8a)

uClibc does not provide res_nsearch:
asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'

Patch coded by Yann E. MORIN:
http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html



ASTERISK-21795 #close

Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
[Retrieved from:
https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch]
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Change-Id: I79296f19e28ec764bbd1e991bf11c416d0b10563
(cherry picked from commit cb0220dec20b988a8d9662dccb215e5a5382b132)

For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.

For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

ASTERISK-30455

Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1
(cherry picked from commit 405211eff757c4e91477d4f6cc6727d3e81057ae)

This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.

ASTERISK-30437 #close
Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6)

* All of the code that used subversion has been removed.

* When Asterisk is checked out from a tag or commit instead
  of one of the regular branches, git would emit messages like
  "fatal: ref HEAD is not a symbolic ref" which weren't fatal
  at all.  Those are now suppressed.

Change-Id: I2a11bc9ebbaf6dfa50f53516ede50a6bac65ca3c
(cherry picked from commit bbec5d1a9926c419265fc2382ce14b37a8ec142e)

Make the existing CURL parameters configurable and allow
to specify the usable protocols, proxy and DNS timeout.

ASTERISK-30340

Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b
(cherry picked from commit 8f088aa0f7ba5a4b955a20ad2854465af3522e84)

Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:

iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
  559 |                         if (!stream->ssl->server) {
      |                                         ^~

ASTERISK-30107 #close

Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587
(cherry picked from commit 030b7b900942df6c795fcabd112d08d1ed858467)

`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:

  ${DAEMON} -rx "core stop now" > /dev/null 2&>1

The intent of which is to execute:

  ${DAEMON} -rx "core stop now"

While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.

So we clean it up and just use bash's shortcut syntax.

Issue raised and a fix suggested (but not used) by peutch on GitHub¹.

ASTERISK-30449 #close

1. https://github.com/asterisk/asterisk/pull/31

Change-Id: Ie279bf4efb4d95cbf507313483d316e977303d19
(cherry picked from commit 46bdd5e3be425279052205e0ad2e688fae87db2a)

There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.

First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.

If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.

This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.

That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)

The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.

To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.

Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.

This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.

ASTERISK-30469

Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7

This reverts commit 56051d1a.

Reason for revert: Behavior change that breaks existing dialplan.

ASTERISK-30472 #close

Change-Id: I83bed3b800d36228a04ded0a6164b795f7f16bd6

* Added a new function ast_utf8_replace_invalid_chars() to
  utf8.c that copies a string replacing any invalid UTF-8
  sequences with the Unicode specified U+FFFD replacement
  character.  For example:  "abc\xffdef" becomes "abc\uFFFDdef".
  Any UTF-8 compliant implementation will show that character
  as a � character.

* Updated res_pjsip:set_id_from_hdr() to use
  ast_utf8_replace_invalid_chars and print a warning if any
  invalid sequences were found during the copy.

* Updated stasis_channels:ast_channel_publish_varset to use
  ast_utf8_replace_invalid_chars and print a warning if any
  invalid sequences were found during the copy.

ASTERISK-27830

Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8

This avoids buffer overflow errors when running tests that capture
output from child processes.

This also corrects a copypasta in an off-nominal error message.

Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d

ASTERISK_29392 (a security fix) introduced a regression by
not processing frames when we don't have an audio format.

Currently, chan_iax2 only calls jb_get to read frames from
the jitterbuffer when the voiceformat has been set on the pvt.
However, this only happens when we receive a voice frame, which
means that prior to receiving voice frames, other types of frames
get stalled completely in the jitterbuffer.

To fix this, we now fallback to using the format negotiated during
call setup until we've actually received a voice frame with a format.
This ensures we're always able to read from the jitterbuffer.

ASTERISK-30354 #close
ASTERISK-30162 #close

Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9

`getcwd(…)` is decorated with the `warn_unused_result` attribute and
therefore needs its return value checked.

Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93

Asterisk makefiles auto-detect SSL library availability,
then they assume that pjproject makefiles will also autodetect
an SSL library at the same time, so they do not pass on the
autodetection result to pjproject.

This normally works, except the pjproject makefiles disables
autodetection when cross-compiling.

Fix by explicitly configuring pjproject to use SSL if we
have been told to use it or it was autodetected

ASTERISK-30424 #close

Change-Id: I8fe2999ea46710e21d1d55a1bed92769c6ebded9

Adds 'e' option to allow Read() to return the terminator as the
dialed digits in the case where only the terminator is entered.

ie; if "#" is entered, return "#" if the 'e' option is set and ""
if it is not.

ASTERISK-30411

Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5

Phones moving between subnets on multi-homed server have their
initially connected interface IP cached in the SERVER variable,
even when it is not specified in the configuration files. This
prevents phones from obtaining the correct SERVER variable value
when they move to another subnet.

ASTERISK-30388 #close
Reported-by: cmaj

Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92

Adds 's' option to skip calling the extension and instead set the
extension as DIRECTORY_EXTEN channel variable.

ASTERISK-30405

Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299

Adds a new option to SendDTMF() which will answer the specified
channel if it is not already up. If no channel is specified, the
current channel will be answered instead.

ASTERISK-30422

Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085

contributed pjproject - patch to check sub->pending_notify
in evsub.c:on_tsx_state before calling
pjsip_evsub_send_request()

res_pjsip_pubsub - change post pjsip 2.13 behavior to use
pubsub_on_refresh_timeout to avoid the ao2_cleanup call on
the sub_tree. This is is because the final NOTIFY send is no
longer the last place the sub_tree is referenced.

ASTERISK-30419

Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438

ASTERISK-30417 #close

Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe

Several queue fields were not being set to their default value during
a reload.

Additionally added some sample configuration options that were missing
from queues.conf.sample.

Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4

Removed multiple patches.

Code chages in res_pjsip_pubsub due to changes in evsub.

Pjsip now calls on_evsub_state() before on_rx_refresh(),
so the sub tree deletion that used to take place in
on_evsub_state() now must take place in on_rx_refresh().

Additionally, pjsip now requires that you send the NOTIFY
from within on_rx_refresh(), otherwise it will assert
when going to send the 200 OK. The idea is that it will
look for this NOTIFY and cache it until after sending the
response in order to deal with the self-imposed message
mis-order. Asterisk previously dealt with this by pushing
the NOTIFY in on_rx_refresh(), but pjsip now forces us
to use it's method.

Changes were required to configure in order to detect
which way pjsip handles this as the two are not
compatible for the reasons mentioned above.

A corresponding change in testsuite is required in order
to deal with the small interal timing changes caused by
moving the NOTIFY send.

ASTERISK-30325

Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f

Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85

Adds the Signal and WaitForSignal
applications, which can be used for inter-channel
signaling in the dialplan.

Signal supports sending a signal to other channels
listening for a signal of the same name, with an
optional data payload. The signal is received by
all channels waiting for that named signal.

ASTERISK-29810 #close

Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3

Adds option to app_directory to specify a filename from which to
read configuration instead of voicemail.conf ie;

same => n,Directory(,,c(directory.conf))

This configuration should contain a list of extensions using the
voicemail.conf format, ie;

2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no

ASTERISK-30404

Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13

Adds support for arrays to JSON_DECODE by allowing the
user to print out entire arrays or index a particular
key or print the number of keys in a JSON array.

Additionally, adds support for recursively iterating a
JSON tree in a single function call, making it easier
to parse JSON results with multiple levels. A maximum
depth is imposed to prevent potentially blowing
the stack.

Also fixes a bug with the unit tests causing an empty
string to be printed instead of the actual test result.

ASTERISK-29913 #close

Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b

Added NULL pointer check and channel lock to prevent resource release
while the chanspy is processing.

ASTERISK-29604

Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24

Variable references within global variable assignments are now
expanded rather than being included literally.

ASTERISK-30406 #close

Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d