Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.

This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.

According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP

The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.

ASTERISK-30193 #close

Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148

Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.

This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it.  Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.

res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.

If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.

ASTERISK-30252

Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb

Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.

This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.

These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.

Current default behavior remains unchanged.

ASTERISK-30091 #close

Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832

This patch adds support for mediasec SIP headers and SDP attributes.
These are defined in RFC 3329, 3GPP TS 24.229 and
draft-dawes-sipcore-mediasec-parameter. The new features are
implemented so that a backbone for RFC 3329 is present to streamline
future work on RFC 3329.

With this patch, Asterisk can communicate with Deutsche Telekom trunks
which require these fields.

ASTERISK-30032

Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2

ASTERISK-30232 #close

Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902

This patch adds a new option to the 100rel parameter for pjsip
endpoints called "peer_supported". When an endpoint with this option
receives an incoming request and the request indicated support for the
100rel extension, then Asterisk will send 1xx responses reliably. If
the request did not indicate 100rel support, Asterisk sends 1xx
responses normally.

ASTERISK-30158

Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad

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

This change allows TEL URI requests to come through for basic calls. The
allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
headers will now allow TEL URIs, as well as the request URI.

Support is only for TEL URIs present in traffic from a remote party.
Asterisk does not generate any TEL URIs on its own.

ASTERISK-26894

Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a

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-30045

Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8

ASTERISK-30037

Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf

ASTERISK-30037

Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2

Use const char for char arguments to
pbx_substitute_variables_helper_full_location
that can do so (context and exten).

ASTERISK-30209 #close

Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f

Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.

Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.

Added a few missing parameters to the ones allowed for writing
with GEOLOC_PROFILE.

Fixed a bug where calling GEOLOC_PROFILE to read a parameter
might actually update the profile object.

Cleaned up XML documentation a bit.

ASTERISK-30190

Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c

You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles.  This is
mutually exclusive with setting location_reference on the
profile.

Updated appdocsxml.dtd to allow xi:include in a configObject
element.  This makes it easier to link to complete configOptions
in another object.  This is used to add the above fields to the
profile object without having to maintain the option descriptions
in two places.

ASTERISK-30185

Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569

Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.

Fixed a possible SEGV if a sub-parameter value didn't have a
value.

ASTERISK-30177

Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd

If "core show channels" is run before startup has completed, it
is possible for bad ao2 refs to occur because the system is not
yet fully initialized. This will lead to an assertion failing.

To prevent this, initialization of CLI builtins is moved to be
later along in the main load sequence. Core CLI commands are
loaded at the same time, but channel-related commands are loaded
later on.

ASTERISK-29846 #close

Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6

This change adds support using the pjsip_tls_transport_restart
function for reloading the TLS certificate and key, if the filenames
remain unchanged. This is useful for Let's Encrypt and other
situations. Note that no restart of the transport will occur if
the certificate and key remain unchanged.

ASTERISK-30186

Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0

The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set()
for us, so no need to declare it separately.

Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a

Adds additional control options over the transfer
feature functionality to give users more control
in how the transfer feature sounds and works.

First, the "transfer" sound that plays when a transfer is
initiated can now be customized by the user in
features.conf, just as with the other transfer sounds.

Secondly, the user can now specify the transfer extension
in advance by using the TRANSFER_EXTEN variable. If
a valid extension is contained in this variable, the call
will automatically be transferred to this destination.
Otherwise, it will fall back to collecting the extension
from the user as is always done now.

ASTERISK-29899 #close

Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3

Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype

ASTERISK-30163 #close

Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7

* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
  case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
  one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
  insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
  reflect it's purpose.
* Added the config option for 'allow_routing_use' which
  sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
  dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
  * Removed the 'profile' argument.
  * Automatically create a profile if it doesn't exist.
  * Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.

ASTERISK-30167

Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0

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

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

* 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

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

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

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

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

ASTERISK-29486

Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432

added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183

ASTERISK-29842

Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18

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

ASTERISK-30021 #close

Change-Id: I70eb59b782a4946b979942e21422746b7563029c

Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.

ASTERISK-29476

Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406

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