* Remove .gitreview and switch to pulling the main asterisk branch
  version from configure.ac instead.

* Replace references to JIRA with GitHub.

* Other minor cleanup found along the way.

Resolves: #39
(cherry picked from commit 5c6d5ea38fd4c4966a91fa8fe4cff417407fff7c)

Fixed CherryPickTest so it triggers only on the
"cherry-pick-test" label instead of all labels.

(cherry picked from commit 9097fc7515d1d281cf8f8b23c5dd76f60cafc232)

(cherry picked from commit fbde0a71f083684f555b559677a85a0e9cd06b59)

(cherry picked from commit aced9046cada3f223e3cd85d89437b130320d7fc)

(cherry picked from commit b7eeef63141e311fda7f04bc530216d87aa4efe7)

When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.

Resolves: #48
(cherry picked from commit f3cc1e7fbd08b1dc90bc7a1c13afae42a7210bf1)

(cherry picked from commit 37071404720203bc68790f59c960bc1880340cdf)

Resolves: #43
(cherry picked from commit a614604f32445509dff3769b8cfa4151645253c0)

Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.

In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.

A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".

Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.

Resolves: #35
ASTERISK-29992

UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.

Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.

(cherry picked from commit 82d7bb49ddc15cff5ea2e50f5ee6934102fe13d1)

(cherry picked from commit ef644c3e937e132c7b91e75f6ca580b04991ab05)

(cherry picked from commit 313cd9dd84521fa30803ae350b2fb6108739f6e6)

(cherry picked from commit 987315a5faa95354ff2d2069898e08ed43ee1d0b)

(cherry picked from commit f8dc0148197dd188c273d1c52c3a6e7f94dd77b3)

ASTERISK-28233

Change-Id: I0f11c096b307a6178e22ca49d9c756343f0e1fdc
(cherry picked from commit 6c53e5e870e393005fa454ec3e6fdce970a95b69)

This reverts commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6.

Reason for revert: Causes segmentation fault.

Change-Id: Ic189c6f7872943a5500d3e71142f0c09d54fcc31
(cherry picked from commit de15852ef01e9512f822656d2c7b27cf4d2678f5)

The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.

RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.

To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.

Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.

ASTERISK-30407 #close

Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
(cherry picked from commit 66a7cff57ecc4514c28071981037afa72ef0d4b9)

Adds PJSIP as a supported technology to DUNDi.

To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.

ASTERISK-28109 #close
ASTERISK-28233 #close

Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf
(cherry picked from commit b33f92cbb56fb848d2a0aaeb416b7cac4813f804)

ASTERISK-30359 #close

Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556
(cherry picked from commit de9aaf7e97c711887d4bf77618677e2c3850d972)

In a three party scenario with INVITE with replaces, we need to
unhold the call, otherwise one party continues to get music on
hold, and the call is not properly bridged between them.

ASTERISK-30428

Change-Id: I5675df11e739be5226b328f8828d4b8d81fbefb4
(cherry picked from commit 1c5720b80226336f62359962850ffd0e5afd2ab7)

A comment at the top of voicemail.conf says that #include
cannot be used in voicemail.conf because this breaks
the ability for app_voicemail to auto-update passwords.
This is factually incorrect, since Asterisk has no problem
updating files that are #include'd in the main configuration
file, and this does work in voicemail.conf as well.

ASTERISK-30479 #close

Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3
(cherry picked from commit d0f39250eeb7e62496477fd4cf57827eb9a99624)

The F option in the xmldocs for the Queue application
was erroneously duplicated, causing it to display
twice on the wiki. The two sections are now merged into one.

Additionally, the description for the d option was quite
vague. Some more details are added to provide context
as to what this actually does.

ASTERISK-30486 #close

Change-Id: I6706cea708b5cc781f59f8652c2cb377e55aed7e
(cherry picked from commit bad5bda08cf4595ee80aa3dca418ccff6356c98c)

The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.

* The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.

* The "failures" attribute was added to the "testsuite"
  element.

Also added 2 new unit tests that just pass and fail to be
used for CI testing.

Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69
(cherry picked from commit a0fd95ef52593e4b7471d8045683f81b101d014a)

Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 019dc51139a2a387fb1dc8aeef038f06afc948b3)

Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe
(cherry picked from commit 6dab013e4991d604146e15f67554b6c1e22c4acc)

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)