The trigger to perform outgoing geolocation processing is the
presence of a geoloc_outgoing_call_profile on an endpoint. This
is intentional so as to not leak location information to
destinations that shouldn't receive it.   In a totally dynamic
configuration scenario however, there may not be any profiles
defined in geolocation.conf.  This makes it impossible to do
outgoing processing without defining a "dummy" profile in the
config file.

This commit adds 4 built-in profiles:
  "<prefer_config>"
  "<discard_config>"
  "<prefer_incoming>"
  "<discard_incoming>"
The profiles are empty except for having their precedence
set and can be set on an endpoint to allow processing without
entries in geolocation.conf.  "<discard_config>" is actually the
best one to use in this situation.

ASTERISK-30182

Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a

When producing an outgoing SDP we iterate through the configured
formats and produce SDP information. It is possible for some
configured formats to not have SDP information available. If this
is the case we skip over them to allow the SDP to still be
produced.

ASTERISK-29185

Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749

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

Fixes two typos that cause fax detection to not work.
One refers to the wrong frame variable, and the other
refers to the subclass.integer instead of the frametype
as it should.

ASTERISK-30192 #close

Change-Id: I7b35fdb7bcf25a29a212eee37c20812c64ab3ef1

Set termination state to old subscriptions to prevent queueing and sending
NOTIFY messages on exten/device state changes.

Postpone destruction of old subscriptions until all already queued tasks
that may be using old subscriptions have completed.

ASTERISK-29906

Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b

* 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

Change-Id: I68ba22db0a69d9e2eabcc2141b48a2395f7f1a23

Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.

ASTERISK-30153 #close

Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b

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

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

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

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

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

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

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

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

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

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

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

As part of PJSIP 2.11 a behavior change was done to require
a matching remote hostname on an established transport for
secure transports. Since the Websocket transport is considered
a secure transport this caused the existing connection to not
be found and used.

We now set the remote hostname and the transport can be found.

ASTERISK-30065

Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94

This is needed to be able to restore it in REGISTER responses,
otherwise the client won't be able to find the contact it created.

ASTERISK-30042

Change-Id: I0c5823918199acf09246b3b206fbde66773688f6

Adjusts the pjsip show registration(s) commands to show
the amount of seconds remaining until a registration
expires.

ASTERISK-29845 #close

Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483

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

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

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