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

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

Three-way calling for analog lines is currently broken.
If party A is on a call with party B and initiates a
three-way call to party C, the behavior differs depending
on whether the call is conferenced prior to party C
answering. The post-answer case is correct. However,
if A flashes before C answers, then the next flash
disconnects B rather than C, which is incorrect.

This error occurs because the subs are not swapped
in the misbehaving case. This is because the flash
handler only swaps the subs if C has answered already,
which is wrong. To fix this, we swap the subs regardless
of whether C has answered or not when the call is
conferenced. This ensures that C is disconnected
on the next hook flash, rather than B as can happen
currently.

ASTERISK-30043 #close

Change-Id: I96c5bf6c9b7eb2636136b716c677c82c079b6f06

Adds an option to VoiceMailMain that prevents the user
from deleting messages during that application invocation.
This can be useful for public or shared mailboxes, where
some users should be able to listen to messages but not
delete them.

ASTERISK-30063 #close

Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26

An m option to Park and ParkAndAnnounce now allows
specifying a music on hold class override.

ASTERISK-30087

Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e

Use example tags instead of regular para tags
where possible.

ASTERISK-30090

Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5

Currently, PJSIP will randomly wait up to 10 seconds for each
outbound registration's initial attempt. The reason for this
is to avoid having all outbound registrations attempt to register
simultaneously.

This can create limitations with the test suite where we need to
be able to receive inbound calls potentially within 10 seconds of
starting up. For instance, we might register to another server
and then try to receive a call through the registration, but if
the registration hasn't happened yet, this will fail, and hence
this inconsistent behavior can cause tests to fail. Ultimately,
this requires a smaller random value because there may be no good
reason to wait for up to 10 seconds in these circumstances.

To address this, a new config option is introduced which makes this
maximum delay configurable. This allows, for instance, this to be
set to a very small value in test systems to ensure that registrations
happen immediately without an unnecessary delay, and can be used more
generally to control how "tight" the initial outbound registrations
are.

ASTERISK-29965 #close

Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd

When a pjsip endpoint is defined with timers=always, this has been a
functional noop.  This patch correctly sets the feature bitmap to both
enable support for session timers and to enable them even when the
endpoint itself does not request or support timers.

ASTERISK-29603
Reported-By: Ray Crumrine

Change-Id: I8b5eeaa9ec7f50cc6d96dd34c2b4aa9c53fb5440

If there is scheduled notification, we must delete it
to avoid using destroyed subscriptions.

ASTERISK-29906

Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f

ASTERISK-29891

Change-Id: Ic8c9697e616446e06e6302653eae902aa23372ad

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 a switch is invoked using chan_iax2, deadlock can result
because the PBX core is autoservicing the channel while chan_iax2
also then attempts to service it while waiting for the result
of the switch. This removes servicing of the channel to prevent
any conflicts.

ASTERISK-30064 #close

Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22

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

res_calendar will trigger an assertion currently
if the ending time is calculated to be in the past.
Unlike the reminder and start times, however, there
is currently no check to catch non-positive times
and set them to 1. As a result, if we get a negative
value by happenstance, this can cause a crash.

To prevent the assertion from begin triggered, we now
use the same logic as the reminder and start events
to catch this issue before it can cause a problem.

ASTERISK-29981 #close

Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c

Emits a warning if the user has requested a parking spot that
is out of bounds for the requested parking lot.

ASTERISK-30086

Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf

When a PJSIP channel is set on hold or off hold, all streams were set
on/off hold. This is not the desired behaviour and caused issues
when there were multiple streams in the topology.

Now, only the default audio stream is set on/off hold when a hold is
indicated.

ASTERISK-30051

Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71

The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local"
Identity Element URI and Target Element URI to the same value -
the channel Caller Number.
For Identity Element it's ok to set as Caller ID.
But Local Target URI should be set as local URI.

In this case the Local Target URI can be used for Directed Call Pickup
by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup).

Also XML sanitized Display names.

ASTERISK-24601

Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343

Agi commnad exec can now evaluate dialplan functions and
variables if variable AGIEXECFULL is set to yes. this can
be useful when executing Playback or Read from agi.

ASTERISK-30058 #close

Change-Id: I669991f540496e7bddd096fec82b52c083036832

Make sure that we have a working sed before trying to use it.

ASTERISK-30059 #close

Change-Id: I9abad67a5df11b665d480feec304ab9d6f55cc76