When ast_func_read2 is used to read a function using
its read function (as opposed to a native ast_str read2
function), the result is copied directly by the function
into the ast_str buffer. As a result, the ast_str length
remains initialized to 0, which is a bug because this is
not the real string length.

This can cascade and have issues elsewhere, such as when
reading substrings of functions that only register read
as opposed to read2 callbacks. In this case, since reading
ast_str_strlen returns 0, the returned substring is empty
as opposed to the actual substring. This has caused
the ast_str family of functions to behave inconsistently
and erroneously, in contrast to the pbx_variables substitution
functions which work correctly.

This fixes this issue by manually updating the ast_str length
when the result is copied directly into the ast_str buffer.

Additionally, an assertion and a unit test that previously
exposed these issues are added, now that the issue is fixed.

ASTERISK-29966 #close

Change-Id: I4e2dba41410f9d4dff61c995d2ca27718248e07f

Adds the DBGetTree action, which can be used to
retrieve all of the DB keys beginning with a
particular prefix, similar to the capability
provided by the database show CLI command.

ASTERISK-30136 #close

Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6

The global event filtering code was only in one
possible execution path, so not all events were
being properly filtered out if requested. This moves
that into the universal AMI handling code so all
events are properly handled.

Additionally, the CLI listing of disabled events can
also get truncated, so we now print out everything.

ASTERISK-30137 #close

Change-Id: If8c42edcb2abc5158552da7eba2a8ff6b20e1959

There are several things wrong with analog Caller ID
handling that are fixed by this commit:

callerid.c's Caller ID generation function contains the
logic to use the presentation to properly send the proper
Caller ID. However, currently, DAHDI does not pass any
presentation information to the Caller ID module, which
means that presentation is completely ignored on all calls.
This means that lines could be getting Caller ID information
they aren't supposed to.

Part of the reason this has been obscured is because the
simple switch logic for handling the built in *67 and *82
is completely wrong. Rather than modifying the presentation
for the call accordingly (which is what it's supposed to do),
it simply blanks out the Caller ID or fills it in. This is
wrong, so wrong that it makes a mockery of the specification.
Additionally, it would leave to the "UNAVAILABLE" disposition
being used for Caller ID generation as opposed to the "PRIVATE"
disposition that it should have been using. This is now fixed
to only update the presentation and not modify the number and
name, so that the simple switch *67/*82 work correctly.

Next, sig_analog currently only copies over the name and number,
nothing else, when it is filling in a duplicated caller id
structure. Thus, we also now copy over the presentation
information so that is available for the Caller ID spill.
Additionally, this meant that "valid" was implicitly 0,
and as such presentation would always fail to "Unavailable".
The validity is therefore also copied over so it can be used
by ast_party_id_presentation.

As part of this fix, new API is added so that all the relevant
Caller ID information can be passed in to the Caller ID generation
functions. Parameters that are also completely missing from the
Caller ID spill have also been added, to enhance the compatibility,
correctness, and completeness of the Asterisk Caller ID implementation.

ASTERISK-29991 #close

Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15

Allocate all of the ast_context's character data in the structure's
flexible array member and eliminate the clunky fake_context. This will
simplify future changes to ast_context.

Change-Id: I98357de75d8ac2b3c4c9f201223632e6901021ea

ASTERISK-30089 #close

Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275

The current documentation is out of date and does not reflect actual
behaviour.  This change makes documentation clearer and accurately
reflect the purpose of relevant channel variables.

ASTERISK-30123

Change-Id: I160d0b01fce862477ad55ac1aa708a730473eb6f

* 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

Currently, if using the CLI to delete a DB entry,
"Database entry removed" is always returned,
regardless of whether or not the entry actually
existed in the first place. This meant that users
were never told if entries did not exist.

The same issue occurs if trying to delete a DB key
using AMI.

To address this, new API is added that is more stringent
in deleting values from AstDB, which will not return
success if the value did not exist in the first place,
and will print out specific error details if available.

ASTERISK-30001 #close

Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18

A corner case exists in CLI parsing where if
a CLI user in a remote console ends with
a backslash and then invokes command completion
(using TAB or ?), then the console will freeze
forever until a SIGQUIT signal is sent to the
process, due to getting blocked forever
reading the command completion. CTRL+C
and other key combinations have no impact on
the CLI session.

This occurs because, in such cases, the CLI
process is waiting for AST_CLI_COMPLETE_EOF
to appear in the buffer from the main process,
but instead the main process is confused by
the funny syntax and thus prints out the CLI help.
As a result, the CLI process is stuck on the
read call, waiting for the completion that
will never come.

This prevents blocking forever by checking
if the data from the main process starts with
"Usage:". If it does, that means that CLI help
was sent instead of the tab complete vector,
and thus the CLI should bail out and not wait
any longer.

ASTERISK-29822 #close

Change-Id: I9810ac59304fec162da701653c9c834f0ec8f670

If the caller has hung up, break out of the play loop so we don't try
to play remaining files and fail to do so.

ASTERISK-30075 #close

Change-Id: I55e85be28ee90b48c0fe4ce20ac136a7dbb49f14

Finding an application and executing it if found is
a common task throughout Asterisk. This adds a helper
function around pbx_exec to do this, to eliminate
redundant code and make it easier for modules to
substitute variables and execute applications by name.

ASTERISK-30061 #close

Change-Id: Ifee4d2825df7545fb515d763d393065675140c84

A previous review fixing ASTERISK_22246 and ASTERISK_26582
got a couple of the options mixed up as to whether or not
they are compatible with the remote console. This fixes
those to the best of my knowledge.

ASTERISK-30097 #close

Change-Id: Id54166991aa79f04fb02699cc499bedda854253b

In function ast_say_date_with_format_de(), take special
care when the hour is one o'clock. In this case, the
German number "eins" must be inflected to its neutrum form,
"ein". This is achieved by playing "digits/1N" instead of
"digits/1". Fixes both 12- and 24-hour formats.

ASTERISK-30092

Change-Id: Ica9b80125c0b317e378d89c1ea786816e2635510

If tab completion using ast_module_helper is attempted
during startup, deadlock will ensue because the CLI
will attempt to lock the module list while it is already
locked by the loader. This causes deadlock because when
the loader tries to acquire the CLI lock, they are blocked
on each other.

Waiting for startup to complete is not feasible because
the CLI lock is acquired while waiting, so deadlock will
ensure regardless of whether or not a lock on the module
list is attempted.

To prevent deadlock, we immediately abort if tab completion
is attempted on the module list before Asterisk is fully
booted.

ASTERISK-30039 #close

Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d

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

ASTERISK-30060 #close

Change-Id: I88d47a1488be2f39017b8d562f993f081844fcb8

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

GCC 12 caught an issue in state_id_by_topic where we were
checking a pointer for NULL instead of the contents of
the pointer for '\0'.

ASTERISK-30044

Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e

When a new unreal (local) channel is created, a second (;2) channel is
created as a counterpart which clones the topology of the first
channel. This creates issues when an outgoing stream is sendonly or
recvonly as the stream state of the inbound channel will be the same
as the stream state of the outbound channel.

Now the stream state is flipped for the streams of the 2nd channel in
ast_unreal_new_channels if the outgoing stream topology is recvonly or
sendonly.

ASTERISK-29655
Reported by: Michael Auracher

ASTERISK-29638
Reported by: Michael Auracher

Change-Id: I0cea29635bb20b7bf7fd0fb95498cd44dab98fbf

Some command line options to Asterisk only apply when Asterisk
is started and cannot be used with remote console mode. If a
user tries to use any of these, they are currently simply
silently ignored.

This prints out a warning if incompatible options are used,
informing users that an option used cannot be used with remote
console mode. Additionally, some clarifications are added to
the help text and man page.

ASTERISK-22246
ASTERISK-26582

Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337

This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.

ASTERISK-29486

Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432

Currently, if a user uses an application like ControlPlayback
to try to rewind a file past the beginning, this can throw
warnings when the file format (e.g. PCM) tries to seek to
a negative offset.

Instead of letting file formats try (and fail) to seek a
negative offset, we instead now catch this in the rewind
function to ensure that we never seek an offset less than 0.
This prevents legitimate user actions from triggering warnings
from any particular file formats.

ASTERISK-29943 #close

Change-Id: Ia53f2623f57898f4b8e5c894b968b01e95426967

Adds the dialplan eval function commands to evaluate a dialplan
function from the CLI. The return value and function result are
printed out and can be used for testing or debugging.

ASTERISK-29820 #close

Change-Id: I833e97ea54c49336aca145330a2adeebfad05209

Adds version information for applications, functions,
and manager events/actions.

This is not completely exhaustive by any means but
covers most new things added that have release
versioning information in the issue tracker.

ASTERISK-29940 #close

Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131

On a write error to an AMI session a flag was set to
indicate that the write error had occurred, with the
expected result being that the session be terminated.
This was not actually happening and instead writing
would continue to be attempted.

This change adds a check for the write error and causes
the session to actually terminate.

ASTERISK-29948

Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5

The PBX core uses the stack when it comes to includes, which
means that a context can only contain strictly fewer than
AST_PBX_MAX_STACK includes. If this is exceeded, then warnings
will be emitted for each number of includes beyond this if
searching for an extension in the including context, and if
the extension's inclusion is beyond the stack size, it will
simply not be found.

To address this, we now check if there are too many includes
in a context when the dialplan is reloaded so that if there
is an issue, the user is aware of at "compile time" as opposed
to "run time" only. Secondly, more details are printed out
when this message is encountered so it's clear what has happened.

ASTERISK-26719

Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958

Treat time_t's as entirely unique and use the POSIX API's for
converting to/from strings.

Lastly, a 64-bit integer formats as 20 digits at most in base10.
Don't need to have any 100 byte buffers to hold that.

ASTERISK-29674 #close

Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56

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

MUSL defines BUFSIZ as 1024 which is not reasonable for log messages.

More broadly, BUFSIZ is the amount of buffering stdio.h does, which
is arbitrary and largely orthogonal to what logging should accept
as the maximum message size.

ASTERISK-29928

Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170

BackGround and WaitExten both accept options that are not
currently documented. This adds documentation for these
options to the xml documentation for each application.

ASTERISK-29967 #close

Change-Id: If812a9f1ccbba3e4d427a0e7a6dea923c2f905f7

ASTERISK_29853 added the ability to selectively disable
AMI events on a global basis, but the logic for this uses
strstr which means that events with names which are the prefix
of another event, if disabled, could disable those events as
well.

Instead, we account for this possibility to prevent this
undesired behavior from occuring.

ASTERISK_29853

Change-Id: Icccd1872602889806740971e4adf932f92466959

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

Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.

ASTERISK-29840 #close

Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307

Although there are 10 debugs levels, over time,
many current debug calls have come to use
inappropriately low debug levels. In particular,
a select few debug calls (currently all debug 1)
can result in thousands of debug messages per minute
for a single call.

This can adds a lot of noise to core debug
which dilutes the value in having different
debug levels in the first place, as these
log messages are from the core internals are
are better suited for higher debug levels.

Some debugs levels are thus adjusted so that
debug level 1 is not inappropriately overloaded
with these extremely high-volume and general
debug messages.

ASTERISK-29897 #close

Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4

Refactors temp file utility function into file.c.

ASTERISK-29809 #close

Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b

Adds two pieces of information to the core show settings command
which are useful in the context of getting backtraces.

The first is to display whether or not Asterisk would generate
a core dump if it were to crash.

The second is to show the current running directory of Asterisk.

ASTERISK-29866 #close

Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83

The configObject tag contains a default attribute which
allows the default value to be specified, if applicable.
This allows for the default value to show up specially on
the wiki in a way that is clear to users.

There are a couple places in the tree where default values
are included in the description as opposed to as attributes,
which means these can't be parsed specially for the wiki.
These are changed to use the attribute instead of being
included in the text description.

ASTERISK-29898 #close

Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755

The disabledevents setting has been added to the general section
in manager.conf, which allows users to specify events that
should be globally disabled and not sent to any AMI listeners.

This allows for processing of these AMI events to end sooner and,
for frequent AMI events such as Newexten which users may not have
any need for, allows them to not be processed. Additionally, it also
cleans up core debug as previously when debug was 3 or higher,
the debug was constantly spammed by "Analyzing AMI event" messages
along with a complete dump of the event contents (often for Newexten).

ASTERISK-29853 #close

Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205