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

Put checks in place to limit how much we will actually download, as well
as a check for the data we receive at the start to ensure it begins with
what we would expect a certificate to begin with.

ASTERISK-29872

Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46

Omit "unsupported column type 'text'" warning in logs while
using text-type column in the PgSQL backend.

ASTERISK-29924 #close

Change-Id: I48061a7d469426859670db07f1ed8af1eb814712

This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.

ASTERISK-29906 #close

Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3

Added the missing xml-stylesheet and Xinclude namespace
declarations in pjsip_config.xml and pjsip_manager.xml.

Updated make_xml_documentation to show detailed errors when
xmlstarlet is the validator.  It's now run once with the '-q'
option to suppress harmless/expected messages and if it actually
fails, it's run again without '-q' but with '-e' to show
the actual errors.

Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e

Moved the xmldoc build logic from the top-level Makefile into
its own script "make_xml_documentation" in the build_tools
directory.

Created a new utility script "get_sourceable_makeopts", also in
the build_tools directory, that dumps the top-level "makeopts"
file in a format that can be "sourced" from shell sscripts.
This allows scripts to easily get the values of common make
build variables such as the location of the GREP, SED, AWK, etc.
utilities as well as the AST* and library *_LIB and *_INCLUDE
variables.

Besides moving logic out of the Makefile, some optimizations
were done like removing "third-party" from the list of
subdirectories to be searched for documentation and changing some
assignments from "=" to ":=" so they're only evaluated once.
The speed increase is noticeable.

The makeopts.in file was updated to include the paths to
REALPATH and DIRNAME.  The ./conifgure script was setting them
but makeopts.in wasn't including them.

So...

With this change, you can now place documentation in any"c"
source file AND you can now place it in a separate XML file
altogether.  The following are examples of valid locations:

res/res_pjsip.c
    Using the existing /*** DOCUMENTATION ***/ fragment.

res/res_pjsip/pjsip_configuration.c
    Using the existing /*** DOCUMENTATION ***/ fragment.

res/res_pjsip/pjsip_doc.xml
    A fully-formed XML file.  The "configInfo", "manager",
    "managerEvent", etc. elements that would be in the "c"
    file DOCUMENTATION fragment should be wrapped in proper
    XML.  Example for "somemodule.xml":

    <?xml version="1.0" encoding="UTF-8"?>
    <!DOCTYPE docs SYSTEM "appdocsxml.dtd">
    <docs>
        <configInfo>
        ...
        </configInfo>
    </docs>

It's the "appdocsxml.dtd" that tells make_xml_documentation
that this is a documentation XML file and not some other XML file.
It also allows many XML-capable editors to do formatting and
validation.

Other than the ".xml" suffix, the name of the file is not
significant.

As a start... This change also moves the documentation that was
in res_pjsip.c to 2 new XML files in res/res_pjsip:
pjsip_config.xml and pjsip_manager.xml.  This cut the number of
lines in res_pjsip.c in half. :)

Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade

Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.

This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.

ASTERISK-29861 #close

Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d

Refactors temp file utility function into file.c.

ASTERISK-29809 #close

Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b

If Subscription refresh occurred between when the batched notification
was scheduled and the serialized notification was to be sent,
then new schedule notification task would never be added.

There are 2 threads:

thread #1. ast_sip_subscription_notify is called,
if notification_batch_interval then call schedule_notification.
1.1. The schedule_notification checks notify_sched_id > -1
not true, then
send_scheduled_notify = 1
notify_sched_id =
  ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
1.2. The sched_cb pushes task serialized_send_notify to serializer
and returns 0 which means no reschedule.
1.3. The serialized_send_notify checks send_scheduled_notify if it's false
the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.

thread #2. pubsub_on_rx_refresh is called
2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
which calls send_notify
2.3. The send_notify set send_scheduled_notify = 0;

The serialized_send_notify should always unset notify_sched_id.

ASTERISK-29904 #close

Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875

Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.

This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.

ASTERISK-29891 #close

Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681

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

added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO
which fixes issue with module crashes on load "FRACK!, Failed assertion"

ASTERISK-29871

Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096

Change-Id: I886936b808def5540d40071321e72f6bfa19063a

In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess
that hasn't been initialized, it'll assert and abort.  If
digest_create_request_with_auth() fails to find the proper
auth object however, it jumps to its cleanup which does exactly
that.  So now we no longer attempt to call pjsip_auth_clt_deinit()
if we never actually initialized it.

ASTERISK-29888

Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62

Fixes some minor logic issues with the module:

Previously, the OPT_END_FILTER flag was getting
tested before options were parsed, so it could
never evaluate to true (wrong ordering).

Additionally, the initially parsed timeout (float)
needs to be compared with 0, not the result int
which is set afterwards (wrong variable).

ASTERISK-29857 #close

Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c

When generating dtmfs, asterisk can incorrectly think packet loss
occured during the dtmf generation, resulting in a jump in sequence
numbers when forwarding voice frames resumes.  This patch forces
asterisk to re-learn the expected sequence number after each DTMF
to avoid this

ASTERISK-29869 #close

Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853

Previously there was no way to specify a connection timeout when
attempting to connect a websocket client to a server. This patch
makes it possible to now do such.

Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d

ASTERISK-28890 #close

Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1

The code currently checks to see if an RFC3389
warning flag is set, except if it is, it merely
sets the flag again, the logic of which doesn't
make any sense.

This adjusts the if comparison to check if the
flag has NOT been set, and if so, emit a notice
log event and set the flag so that future frames
do not cause an event to be logged.

ASTERISK-29856 #close

Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1

Neither pjsip_message_filter's filter_on_tx_message() nor
res_pjsip_session's session_outgoing_nat_hook() were multipart
aware and just assumed that an SDP would be the only thing in
a message body.  Both were changed to use the new
pjsip_get_sdp_info() function which searches for an sdp in
both single- and multi- part message bodies.

ASTERISK-29813

Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56

Added two new functions to assist checking media types...

* ast_sip_are_media_types_equal compares two pjsip_media_types.
* ast_sip_is_media_type_in tests if one media type is in a list
  of others.

Added static definitions for commonly used media types to
res_pjsip.h.

Changed several modules to use the new functions and static
definitions.

ASTERISK_29813
(not ready to close)

Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9

There are times when you need to troubleshoot issues with bundled
pjproject or add new features that need to be pushed upstream
but...

* The source directory created by extracting the pjproject tarball
  is not scanned for code changes so you have to keep forcing
  rebuilds.
* The source directory isn't a git repo so you can't easily create
  patches, do git bisects, etc.
* Accidentally doing a make distclean will ruin your day by wiping
  out the source directory, and your changes.
* etc.

This commit makes that easier.
See third-party/pjproject/README-hacking.md for the details.

ASTERISK-29824

Change-Id: Idb1251040affdab31d27cd272dda68676da9b268

The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.

ASTERISK-28863
ASTERISK-29320

Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25

Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016

res_sdp_crypto_parse_offer(.) emits many log messages already.

ASTERISK-29785

Change-Id: I1a191ebe4fec1102946d4e31887e5197ca02dfe8

res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
that rtpstart was configured to be an odd value. Also adding a loop
counter to prevent a possible infinite loop when looking for a free
port.

ASTERISK-27406

Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b

These are the remaining issues found in /res.

ASTERISK-29761

Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d

Newer versions of spandsp did refactoring of code to add new features
like color FAXing. This refactoring broke backwards compatibility.
Add support for the new version while retaining support for 0.0.6.

ASTERISK-29729 #close

Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1

ASTERISK-29763

Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e

ASTERISK-29779

Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.

This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.

ASTERISK-29777 #close

Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b

ASTERISK-29776

Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73

Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.

ASTERISK-29720 #close

Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90

ASTERISK-29754

Change-Id: Ia09eb68d283d201d9a6fbeccfc0efe83fe0502a5

ASTERISK-29753

Change-Id: I7a61974584f6169502e6860fc711919fe7bbfaa7

ASTERISK-29756

Change-Id: I2f1c1eea1c902492b77b74de9950f20ebbb7e758

ASTERISK-29750

Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9

ASTERISK-29749

Change-Id: I7885793b63bdeaa883e76edb899bbba9660eb1c5

ASTERISK-29747

Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084

Fixes four misuses of the parameter 'name'. Additionally, for
consistency and to avoid such an issue in future, those few other
places, which used '\file name', were changed just to '\file'. Then,
Doxygen uses the name of the current file.

ASTERISK-29733

Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d