This reverts commit 3c8116c8. It has
been a merge that has been acidentally squashed into one commit with the
consequence that all history has been lost. The next commit will be a
proper merge commit rectifying this.

Grzegorz Sluja authored 1 year ago and Yalu Zhang committed 1 year ago

Our commits to the previous version have been rebased.

* 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)

This reverts commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6.

Reason for revert: Causes segmentation fault.

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

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)

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)

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)

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

ASTERISK-30440 #close

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

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)

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)

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

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 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

Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.

ASTERISK-30262 #close

Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db

In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b3) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.

Partial fix for ASTERISK~25661.

Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff

Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).

ASTERISK-30319 #close

Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645

-----------------

This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures.  The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.

ASTERISK-30391

-----------------

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038

This reverts commit d454801c.

Reason for revert: Issue when transcoding to/from g722

Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86

Adds a new application, Broadcast, which can be used for
one-to-many transmission and many-to-one reception of
channel audio in Asterisk. This is similar to ChanSpy,
except it is designed for multiple channel targets instead
of a single one. This can make certain kinds of audio
manipulation more efficient and streamlined. New kinds
of audio injection impossible with ChanSpy are also made
possible.

ASTERISK-30180 #close

Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

ASTERISK-30280

Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a

ASTERISK-21502

Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08

chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).

This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.

* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
  INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)

The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.

The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.

Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.

ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>

Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333

msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.

This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.

ASTERISK-30283 #close

Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7

Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.

ASTERISK-30322 #close

Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b

Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.

ASTERISK-29497

Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49

Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.

ASTERISK-30150 #close

Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428

The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).

This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.

ASTERISK-30289 #close

Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07

MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.

ASTERISK-30286 #close

Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c

Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.

ASTERISK-30176

Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404

The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.

This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.

ASTERISK-30308 #close

Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4

Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5

Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.

This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.

ASTERISK-30284 #close

Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e