There are a handful of files in the tree that
reference an SVN link for the coding guidelines.

This removes these because the links are dead
and the vast majority of source files do not
contain these links, so this is more consistent.

app_skel still maintains an (up to date) link
to the coding guidelines.

ASTERISK-30159 #close

Change-Id: I35bbb20f66982e98099cff3029ede20091ffdac7

Documents the ConfbridgeListRooms AMI response,
which is currently not documented.

ASTERISK-30020 #close

Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea

The MeetmeList and MeetmeListRooms AMI
responses are currently completely undocumented.
This adds documentation for these responses.

ASTERISK-30018 #close

Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8

Adds missing documentation for the field parameter
for the SRVRESULT function.

ASTERISK-30151
Reported by: Chris Young

Change-Id: I4385a2e0892a07e30dea1a8a0588e2c1bea2b1f1

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

configure script detects /sbin/launchd, but the result of this
check is not used in Makefile (bininstall). Makefile also detects
/sbin/launchd file to decide if it is required to install
safe_asterisk.

configure script correctly detects cross compile build and sets
PBX_LAUNCHD=0

In case of building asterisk on MacOS host for Linux target using
external toolchain (e.g. OpenWrt toolchain), bininstall does not
install safe_asterisk (due to /sbin/launchd detection in Makefile),
but it is required on target (Linux).

This patch adds HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@ to makeopts.in to
use the result of /sbin/launchd detection from configure script in
Makefile.
Also this patch uses HAVE_SBIN_LAUNCHD in Makefile (bininstall) to
decide if it is required to install safe_asterisk.

ASTERISK-29905 #close

Change-Id: Iff61217276cd188f43f51ef4cdbffe39d9f07f65

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

Change-Id: Ib91db9223a78188667e15053bcc73931b878414e

Move the call to ast_sip_location_prune_boot_contacts() *after* the call
to ast_res_pjsip_init_options_handling() so that
res/res_pjsip/pjsip_options.c is informed about the contact deletion and
updates its sip_options_contact_statuses list. This allows for an AMI
event to be sent by res/res_pjsip/pjsip_options.c if the endpoint
registers again from the same remote address and port (i.e., same URI)
as used before the Asterisk restart.

ASTERISK-30109
Reported-by: Michael Neuhauser

Change-Id: I1ba4478019e4931a7085f62708d9b66837e901a8

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

ASTERISK-30126 #close

Change-Id: I009c4dcbf9338a13e3baf87b52a5bbe4f9f81a42

Adds a POLARITY function which can be used to
retrieve the current polarity of an FXS channel
as well as set the polarity of an FXS channel
to idle or reverse at any point during a call.

ASTERISK-30000 #close

Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1

make_version now silently checks if the required git commands will
fail.  If they do, then return UNKNOWN__git_check_fail to
distinguish this failure from other UNKNOWN__ version failures

Makefile checks for this value on install and exits out with
instructions

ASTERISK-30029

Change-Id: If8f10cac8f509c08981120f17555762342020221

Currently, if multiple video-enabled ConfBridges are
conferenced together, we immediately get into a scenario
where an infinite sequence of video updates fills up
the taskprocessor queue and causes memory consumption
to climb unabated until Asterisk is killed. This is due
to the core bridging mechanism that provides video updates
(softmix_bridge_write_control in bridge_softmix.c)
continously updating all the channels in the bridge with
video updates.

The logic to do so in the core is that the video updates
should be provided if the video_update_discard property
for the bridge is 0, or if enough time has elapsed since
the last video update. Thus, we already have a safeguard
built in to ensure the scenario described above does not
happen. Currently, however, this safeguard is not being
adequately ensured.

In app_confbridge, the video_update_discard property
defaults to 2000, which is a healthy value that should
completely prevent this issue. However, this value is
only set onto the bridge in the SFU video mode. This
leaves video modes such as follow_talker completely
vulnerable, since video_update_discard will actually
be 0, since the default or set value was never applied.
As a result, the core bridging mechanism will always
try to provide video updates regardless of when the last
one was sent.

To prevent this issue from happening, we now always
set the video_update_discard property on the bridge
with the value from the bridge profile. The app_confbridge
defaults will thus ensure that infinite video updates
no longer happen in any video mode.

ASTERISK-29907 #close

Change-Id: I4accb2536ac62797950468e9930f12ef7dd486b2

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

line 196:    loc_src = '\0';
should have been
line 196:    *loc_src = '\0';

The issue was caught by the gcc optimizer complaining that
loc_src had a zero length because the pointer itself was being
set to NULL instead of the _contents_ of the pointer being set
to the NULL terminator.

ASTERISK-30138
Reported-by: Sean Bright

Change-Id: Id247be113cc8510f043ca053d5b4f5f3d32acd29

This commit adds res_pjsip_geolocation which gives chan_pjsip
the ability to use the core geolocation capabilities.

This commit message is intentionally short because this isn't
a simple capability.  See the documentation at
https://wiki.asterisk.org/wiki/display/AST/Geolocation
for more information.

THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
USER FEEDBACK!

ASTERISK-30128

Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9

This commit adds res_geolocation which creates the core capabilities
to manipulate Geolocation information on SIP INVITEs.

An upcoming commit will add res_pjsip_geolocation which will
allow the capabilities to be used with the pjsip channel driver.

This commit message is intentionally short because this isn't
a simple capability.  See the documentation at
https://wiki.asterisk.org/wiki/display/AST/Geolocation
for more information.

THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
USER FEEDBACK!

ASTERISK-30127

Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303

ASTERISK-30089 #close

Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275

See also: ASTERISK_30023

ASTERISK-30096 #close
patches:
  inline on issue - submitted by Morvai Szabolcs

Change-Id: I79c0b74862100acd9c8319dca5cc456a654d02eb

ASTERISK_30007 accidentally made OpenSSL a
required depdendency. This adds an ifdef so
the relevant code is compiled only if OpenSSL
is available, since it only needs to be executed
if OpenSSL is available anyways.

ASTERISK-30083 #close

Change-Id: Iad05c1a9a8bd2a48e7edf8d234eaa9f80779e34d

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

Adds a CLI command similar to "dialplan eval function" except for
applications: "dialplan exec application", useful for quickly
testing certain application behavior directly from the CLI
without writing any dialplan.

ASTERISK-30062 #close

Change-Id: I42e9fa9b60746c21450d40f99a026d48d2486dde

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

Improved variable definitions (specified correct type) for avoiding
compiler warnings.

ASTERISK-30117 #close

Change-Id: I3b00c1befb658ee9379ddabd9a9132765ca9201a

These new functions allow retrieving information from headers on 200 OK
INVITE response.

ASTERISK-29999

Change-Id: I264a610a9333359297a0825feb29a1bb4f4ad144

Switched res_pjsip_outbound_registration.so dep to optional. Added
module loaded check before using it.

ASTERISK-30101 #close

Change-Id: Ia34f1684d984e821fbdd4de8911f930337703666

ASTERISK_28638 caused a regression by incorrectly aborting
early and overwriting the status on certain calls.
This was exhibited by certain technologies such as DAHDI,
where DAHDI returns NULL for the request if a line is busy.
This caused the BUSY condition to be incorrectly treated
as CHANUNAVAIL because the DIALSTATUS was getting incorrectly
overwritten and call handling was aborted early.

This is fixed by instead checking if any valid peers have been
specified, as opposed to checking the list size of successful
requests. This is because the latter could be empty but this
does not indicate any kind of problem. This restores the
previous working behavior.

ASTERISK-29989 #close

Change-Id: I4d4b209b967816b1bc791534593ababa2b99bb88

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

The Dial application currently stops hook flashes
dead in their tracks from propagating through on
outbound calls. This fixes that so they can go
down the wire.

ASTERISK-30115 #close

Change-Id: Id4e78b29a049f35c5b1e7520eaa10d0eb5b7f97c

Microsoft recently began rejecting all requests for
ICS calendars on Office 365 with 400 errors if
the request doesn't contain a user agent. See:

https://docs.microsoft.com/en-us/answers/questions/883904/34the-remote-server-returned-an-error-400-bad-requ.html

Accordingly, we now send a user agent on requests for
ICS files so that requests to Office 365 will work as
they did before.

ASTERISK-30106

Change-Id: Ie9dcaef12ae8adf37533c684499eb11005fac8f7

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

Rightly the use of wildcards in certificates is disallowed in accordance
with RFC5922. However, RFC2818 does make some allowances with regards to
their use when using subject alt names with DNS name types.

As such this patch creates a new setting for TLS transports called
'allow_wildcard_certs', which when it and 'verify_server' are both enabled
allows DNS name types, as well as the common name that start with '*.'
to match as a wildcard.

For instance: *.example.com
will match for: foo.example.com

Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc...
And the starting wildcard only matches for a single level.

For instance: *.example.com
will NOT match for: foo.bar.example.com

The new setting is disabled by default.

ASTERISK-30072 #close

Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4

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

More information:
https://github.com/pjsip/pjproject/releases/tag/2.12.1

Pull request to third-party
https://github.com/asterisk/third-party/pull/11

ASTERISK-30050

Change-Id: Icb4e86d4b85ef9b975355c91f3ed56a50b51c6bd

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