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

Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93

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

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

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

Removes the function mkstemp_file and uses
ast_file_mkftemp from file.h instead.

ASTERISK-30295 #close

Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609

Fixed the specification of "outputdir" when calling ast_coredumper
so the txt files are saved in the correct place.

ASTERISK-30282

Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6

Fix compilation errors caused by using size_t
instead of uintmax_t and non-portable format
specifiers.

ASTERISK-30273 #close

Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee

ASTERISK-30213 #close

Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a

ASTERISK-30232 #close

Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902

ASTERISK-30235

Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd

Fixed a segfault caused by var_list_from_loc_info() encountering
an empty location info element.

Fixed an issue in ast_strsep() where a value with only whitespace
wasn't being preserved.

Fixed an issue in ast_variable_list_from_quoted_string() where
an empty value was considered a failure.

ASTERISK-30215
Reported by: Dan Cropp

Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a

ASTERISK-30046 #close

Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2

We're validating the following functionality:

encrypting a block of data with RSA
decrypting a block of data with RSA
signing a block of data with RSA
verifying a signature with RSA
encrypting a block of data with AES-ECB
encrypting a block of data with AES-ECB

as well as accessing test keys from the keystore.

ASTERISK-30045 #close

Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d

ASTERISK-30037 #close

Change-Id: I0273e85eeeb6b8e46703f24cd74d84f3daf0a69a

ASTERISK-26826

Change-Id: I07388d16f74452cebc9c981f99044eb6b77df792

Requires Python3 testsuite changes

ASTERISK-26826

Change-Id: I92ec7dec751ad455503a584d6e860db88c56d6bc

A sporadic test failure was happening when executing the AEAP
Websocket transport tests. It was originally thought this was
due to things not getting cleaned up fast enough, but upon further
investigation I determined the underlying cause was poll()
getting interrupted and this not being handled in all places.

This change adds EINTR and EAGAIN handling to the Websocket
client connect code as well as the AEAP Websocket transport code.
If either occur then the code will just go back to waiting
for data.

The originally disabled failure test case has also been
re-enabled.

ASTERISK-30099

Change-Id: I1711a331ecf5d35cd542911dc6aaa9acf1e172ad

* Added ast_variable_list_from_quoted_string()
  Parse a quoted string into an ast_variable list.

* Added ast_str_substitute_variables_full2()
  Perform variable/function/expression substitution on an ast_str.

* Added ast_strsep_quoted()
  Like ast_strsep except you can specify a specific quote character.
  Also added unit test.

* Added ast_xml_find_child_element()
  Find a direct child element by name.

* Added ast_xml_doc_dump_memory()
  Dump the specified document to a buffer

* ast_datastore_free() now checks for a NULL datastore
  before attempting to destroy it.

Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec

The 'transport_binary' test sporadically fails, but on a theory that the
problem is caused by a previously executed test, transport_connect_fail,
part of that test has been disabled until a solution is found.

ASTERISK_30099

Change-Id: I48ed74d696aa9b6159f59661f3d535cac4c909e1

This change exposes the channel driver's unique id (i.e. the Call-ID
for chan_sip/chan_pjsip based channels) to ARI channel resources
as `protocol_id`.

ASTERISK-30027
Reported by: Moritz Fain
Tested by: Moritz Fain

Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87

Most issues were in stringfields and had to do with comparing
a pointer to an constant/interned string with NULL.  Since the
string was a constant, a pointer to it could never be NULL so
the comparison was always "true".  gcc now complains about that.

There were also a few issues where determining if there was
enough space for a memcpy or s(n)printf which were fixed
by defining some of the involved variables as "volatile".

There were also a few other miscellaneous fixes.

ASTERISK-30044

Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570

Add framework to connect to, and read and write protocol based
messages from and to an external application using an Asterisk
External Application Protocol (AEAP). This has been divided into
several abstractions:

 1. transport - base communication layer (currently websocket only)
 2. message - AEAP description and data (currently JSON only)
 3. transaction - links/binds requests and responses
 4. aeap - transport, message, and transaction handler/manager

This patch also adds an AEAP implementation for speech to text.
Existing speech API callbacks for speech to text have been completed
making it possible for Asterisk to connect to a configured external
translator service and provide audio for STT. Results can also be
received from the external translator, and made available as speech
results in Asterisk.

Unit tests have also been created that test the AEAP framework, and
also the speech to text implementation.

ASTERISK-29726 #close

Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2

Removes some leftover build and config references to
modules that have since been removed from Asterisk.

ASTERISK-29935 #close

Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af

Passing 0 as the last argument to strtoimax() or strtoumax() causes
octal and hexadecimal to be accepted which was not originally
intended. So we now force to only accept decimal.

ASTERISK-29950 #close

Change-Id: I93baf0f273441e8280354630a463df263a8c0edd

Added functions to open, close, and apply XML Stylesheets
to XML documents.  Although the presence of libxslt was already
being checked by configure, it was only happening if xmldoc was
enabled.  Now it's checked regardless.

Added ability to parse a string consisting of comma separated
name/value pairs into an ast_variable list.  The reverse of
ast_variable_list_join().

Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e

Added:

Replace a variable in a list:
int ast_variable_list_replace_variable(struct ast_variable **head,
    struct ast_variable *old, struct ast_variable *new);
Added test as well.

Create a "name=value" string from a variable list:
'name1="val1",name2="val2"', etc.
struct ast_str *ast_variable_list_join(
    const struct ast_variable *head, const char *item_separator,
    const char *name_value_separator, const char *quote_char,
    struct ast_str **str);
Added test as well.

Allow the name of an XML element to be changed.
void ast_xml_set_name(struct ast_xml_node *node, const char *name);

Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb

Recap from earlier commit:  If you have a development branch for a
major project that will receive gerrit reviews it'll probably be
named something like "development/16/newproject" or a work branch
based on that "development" branch.  That will necessitate
setting "defaultbranch=development/16/newproject" in .gitreview.
The make_version script uses that variable to construct the
asterisk version however, which results in versions
like "GIT-development/16/newproject-ee582a8c7b" which is probably
not what you want.  It also constructs the URLs for downloading
external modules with that version, which will fail.

Fast-forward:

The earlier attempt at adding a "basebranch" variable to
.gitreview didn't work out too well in practice because changes
were made to .gitreview, which is a checked-in file.  So, if
you wanted to rebase your work branch on the base branch, rebase
would attempt to overwrite your .gitreview with the one from
the base branch and complain about a conflict.

This is a slighltly different approach that adds three methods to
determine the mainline branch:

1.  --- MAINLINE_BRANCH from the environment

If MAINLINE_BRANCH is already set in the environment, that will
be used.  This is primarily for the Jenkins jobs.

2.  --- .develvars

Instead of storing the basebranch in .gitreview, it can now be
stored in a non-checked-in ".develvars" file and keyed by the
current branch.  So, if you were working on a branch named
"new-feature-work" based on "development/16/new-feature" and wanted
 to push to that branch in Gerrit but wanted to pull the external
 modules for 16, you'd create the following .develvars file:

[branch "new-feature-work"]
    mainline-branch = 16

The .gitreview file would still look like:

[gerrit]
defaultbranch=development/16/new-feature

...which would cause any reviews pushed from "new-feature-work" to
go to the "development/16/new-feature" branch in Gerrit.

The key is that the .develvars file is NEVER checked in (it's been
added to .gitignore).

3.  --- Well Known Development Branch

If you're actually working in a branch named like
"development/<mainline_branch>/some-feature", the mainline branch
will be parsed from it.

4.  --- .gitreview

If none of the earlier conditions exist, the .gitreview
"defaultbranch" variable will be used just as before.

Change-Id: I1cdeeaa0944bba3f2e01d7a2039559d0c266f8c9

Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.

This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.

ASTERISK-29861 #close

Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d

Adds a new option, defaultenabled, to the CDR core to
control whether or not CDR is enabled on a newly created
channel. This allows CDR to be disabled by default on
new channels and require the user to explicitly enable
CDR if desired. Existing behavior remains unchanged.

ASTERISK-29808 #close

Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1

sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an
executing call-back. This is done by adding a new variable 'rescheduled'
to the struct sched which is set in ast_sched_runq and checked in
ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for
now deprecated ast_sched_del which returns a new possible value -2
if called on an executing call-back with rescheduled set. ast_sched_del
is modified to call ast_sched_del_nonrunning to maintain existing code.
AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it
will not throw a warning or invoke refcall.
test_sched: Add a new unit test sched_test_freebird that will check the
reference count in the resolved scenario.

ASTERISK-29698

Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d

When test_timezone_watch runs very near a DST transition,
two time zones that would otherwise be expected to report the same
time can differ because of the DST transition.

Instead of having the test fail when this happens, report the
times, time zones, and dst flags.

ASTERISK-29722

Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a

Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

ASTERISK-29779

Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

ASTERISK-29741

Change-Id: I012d72b237bda2ef2d0f86307dfc6dc7add4b54b

Correct typos of the following word families:

mounting
jitterbuffer
thrashing
original
manipulating
entries
actual
possibility
tasks
options
positives
taskprocessor
other
dynamic
declarative

ASTERISK-29714

Change-Id: I6b94659d045eec5d8d020fce2e9b6e2f593dfeb6

Jenkins renamed the 'master' node to 'built-in' in version
2.319 so we have to adjust as well.

Change-Id: Ice663c3a66d0eedf76e8e5fe530328455991ec25

Add a function to check if there is an exact match a one string between
delimiters in another string.

Add a function that will create an ast_json object out of a list of
Asterisk variables. An excludes string can also optionally be passed
in.

Also, add a macro to make it easier to get object integers.

Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226

test_voicemail_api: Use empty char* for empty_msg_ids.
chan_skinny: Fix size of calledParty to be maximum extension.
menuselect: Change Makefile to stop deprecated warnings. Added comments
test_linkedlist: 'bogus' variable was manually allocated from a macro
and the test fails if this happens but the compiler couldn't 'see' this
and returns a warning. memset to all 0's after allocation.
chan_ooh323: Fixed various indentation issues that triggered misleading
 indentation warnings.

ASTERISK-29682
Reported by: George Joseph

Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe

Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c

We can't rely on RAII_VAR(...) to properly clean up data that is
allocated within a loop.

ASTERISK-27176 #close

Change-Id: Ib575616101230c4f603519114ec62ebf3936882c