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

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

Rounding issues with double math were causing rtp timestamp
slips in outgoing packets.  We're now back to integer math
and are getting no more slips.

ASTERISK-30391

Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4

Each playback of WAV files results in logging
"Skipping unknown block 'LIST'".

To prevent unnecessary flooding of this DEBUG log this patch replaces
ast_log(LOG_DEBUG, ...) by ast_debug(1, ...).

Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0

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

Currently, if a module declines to load, dlopen is called
to register the module but dlclose never gets called.
Furthermore, loader.c currently doesn't allow dlclose
to ever get called on the module, since it declined to
load and the unload function bails early in this case.

This can be problematic if a module is updated, since the
new module cannot be loaded into memory since we haven't
closed all references to it. To fix this, we now allow
modules to be unloaded, even if they never "loaded" in
Asterisk itself, so that dlclose is called and the module
can be properly cleaned up, allowing the updated module
to be loaded from scratch next time.

ASTERISK-30345 #close

Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5

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

Since text frames contain a text body, make FRAME_TRACE
more useful for text frames by actually printing the text.

ASTERISK-30353 #close

Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50

json.h contains macros to get a string and an integer
from a JSON object. However, the macro to do this for
JSON reals is missing. This adds that.

ASTERISK-30361 #close

Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a

The if statement here is always false after the for
loop finishes, so variables are never appended.
This removes that to properly append to the end
of the variable list.

ASTERISK-30351 #close
Reported by: Sebastian Gutierrez

Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199

When Asterisk receives a new websocket conenction, it creates a new
pjsip transport for it and copies connection data into it.  The
transport manager then uses the remote IP address and port on the
transport to create a monitor for each connection.  However, the
remote port wasn't being copied, only the IP address which meant
that the transport manager was creating only 1 monitoring entry for
all websocket connections from the same IP address. Therefore, if
one of those connections failed, it deleted the transport taking
all the the connections from that same IP address with it.

* We now copy the remote port into the created transport and the
  transport manager behaves correctly.

ASTERISK-30369

Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1

If native HTTP is disabled but HTTPS is enabled and status page enabled
too, Core/HTTP crashes while loading. 'global_http_server' references
to NULL, but the status page tries to dereference it.

The patch adds a check for HTTP is enabled.

ASTERISK-30379 #close

Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249

Do not crash when a URL has no path component as in this case the
ast_uri_path function will return NULL. Make the code cope with not
having a path.

The below would crash
> media cache create http://google.com /tmp/foo.wav

Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault.
0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
(gdb) bt
 #0  0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
 #1  0x0000ffff43d43a78 in file_extension_from_string (str=<optimized out>, buffer=buffer@entry=0xffffca9973c0 "",
    capacity=capacity@entry=64) at res_http_media_cache.c:288
 #2  0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568,
    buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378
 #3  0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392
 #4  0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555
 #5  0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
    at res_http_media_cache.c:613
 #6  0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
    at bucket.c:191
 #7  0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718,
    details=details@entry=0xffffca9974a8) at sorcery.c:2027
 #8  0x0000000000559698 in ast_sorcery_create (sorcery=<optimized out>, object=object@entry=0x3bf96568) at sorcery.c:2077
 #9  0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727
 #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com",
    file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335
 #11 0x00000000004f88ec in media_cache_handle_create_item (e=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8)
    at media_cache.c:640

ASTERISK-30375 #close

Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273

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

pbx_exec makes a channel snapshot before executing applications.
This doesn't cause an issue during normal dialplan execution
where pbx_exec is called over and over again in succession.
However, if pbx_exec is called "one off", e.g. using
ast_pbx_exec_application, then a channel snapshot never ends
up getting made after the executed application returns, and
inaccurate snapshot information will linger for a while, causing
"core show channels", etc. to show erroneous info.

This is fixed by manually making a channel snapshot at the end
of ast_pbx_exec_application, since we anticipate that pbx_exec
might not get called again immediately.

ASTERISK-30367 #close

Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086

Currently, there is no Caller ID available to us when
checking for an extension match when handling INVITEs.
As a result, extension patterns that depend on the Caller ID
are not matched and calls may be incorrectly rejected.

The Caller ID is not available because the supplement that
adds Caller ID to the session does not execute until after
this check. Supplement callbacks cannot yet be executed
at this point since the session is not yet in the appropriate
state.

To fix this without impacting existing behavior, the Caller ID
number is now retrieved before attempting to pattern match.
This ensures pattern matching works correctly and there is
no behavior change to the way supplements are called.

ASTERISK-28767 #close

Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7

When a call is put on hold and it has moh_passthrough and rtp_timeout
set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is
expected to be used, but rtp_timeout is used instead. This change adds a
couple of checks for locally_held to determine if rtp_timeout_hold needs
to be used instead of rtp_timeout.

ASTERISK-30350

Change-Id: I7b106fc244332014216d12bba851cefe884cc25f

Fixes a negative offset warning by initializing
the buffer to empty.

Additionally, although it doesn't currently complain
about it, the size of a buffer is increased to
accomodate the maximum size contents it could have.

ASTERISK-30240 #close

Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877

Currently, if a user attempts to set a Caller ID related
function to an invalid value, a warning is emitted,
except for when setting the redirecting reason.
We now emit a warning if we were unable to successfully
parse the user-provided reason.

ASTERISK-30332 #close

Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a

Fix aor lookup on sip path addition. Issue happens in case of dialing
with @ and overriding user part of RURI.

ASTERISK-30100 #close
Reported-by: Yury Kirsanov

Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13

When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.

This change ensures that the stream is closed in case of error.

ASTERISK-30198 #close
Reported-by: Julien Alie

Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803

Removes see-also references to applications that don't
exist anymore (removed in Asterisk 19),
so these dead links don't show up on the wiki.

ASTERISK-30347 #close

Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909

The `ast_geoloc_datastore_add_eprofile` function does not return 0 on
success, it returns the size of the underlying datastore. This means
that the datastore will be freed and its pointer set to NULL when no
error occured at all.

ASTERISK-30346

Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9

When adding AOC to an outgoing response the code
assumed that a body would exist for comparing the
Content-Type. This isn't always true.

The code now checks to make sure the response has
a body before checking the Content-Type.

ASTERISK-21502

Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76

Fixes format truncation warnings in gcc 12.2.1.

ASTERISK-30349 #close

Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c

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

When passing a JSON body to the 'create' channel route
it would be converted into Asterisk variables, but never
freed resulting in a memory leak.

This change makes it so that the variables are freed in
all cases.

ASTERISK-30344

Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d

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

This fixes a small typo in the from_domain documentation on the endpoint documentation

ASTERISK-30328 #close

Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21

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

Some SIP devices use an empty extension for PLAR functionality.

Rather than rejecting these empty extensions, we now use the s
extension for such calls to mirror the existing PLAR functionality
in Asterisk (e.g. chan_dahdi).

ASTERISK-30265 #close

Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590

Updates the documentation for the 'contact_user' field to point out the
default outbound contact if no contact_user is specified 's'

ASTERISK-30316 #close

Change-Id: I61f24fb9164e4d07e05908a2511805281874c876