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

Fix parsing of ANI2/OLI information, since it was previously
parsing the user, when it should have been parsing other_param.

Also improves the parsing by using pjproject native functions
rather than trying to parse the parameters ourselves like
chan_sip did. A previous attempt at this caused a crash, but
this works correctly now.

ASTERISK-29703 #close

Change-Id: I8f3c79032d9ea1a21d16f8e11f22bd8d887738a1

Correct typos of the following word families:

identifying
structures
actcount
initializer
attributes
statement
enough
locking
declaration
userevent
provides
unregister
session
execute
searches
verification
suppressed
prepared
passwords
recipients
event
because
brief
unidentified
redundancy
character
the
module
reload
operation
backslashes
accurate
incorrect
collision
initializing
instance
interpreted
buddies
omitted
manually
requires
queries
generator
scheduler
configuration has
owner
resource
performed
masquerade
apparently
routable

ASTERISK-29714

Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda

The stir_shaken configuration option now has 4 different choices to pick
from: off, attest, verify, and on. Off and on behave the same way they
do now. Attest will only perform attestation on the endpoint, and verify
will only perform verification on the endpoint.

Certain responses are required to be sent based on certain conditions
for STIR/SHAKEN. For example, if we get a Date header that is outside of
the time range that is considered valid, a 403 Stale Date response
should be sent. This and several other responses have been added.

Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7

Add a new function that converts a speech results type to a string.
Also add another function to unregister an engine, but returns a
pointer to the unregistered engine object instead of a success/fail
integer.

Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192

In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.

ASTERISK-29402

Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557

Add missing reference decrement in rtp_deallocate_transport()

ASTERISK-29671

Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9

The behavior of max_contacts and remove_existing are connected.  If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact.  Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.

This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing.  If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.

ASTERISK-29525

Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784

When listing bridges we go through the ones present in
ARI, get their snapshot, turn it into JSON, and add it
to the payload we ultimately return.

An invisible "dial bridge" exists within ARI that would
also try to be added to this payload if the channel
"create" and "dial" routes were used. This would ultimately
fail due to invisible bridges having no snapshot
resulting in the listing of bridges failing.

This change makes it so that the listing of bridges
ignores invisible ones.

ASTERISK-29668

Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a

Rather than stripping parameters from Content-Type headers before
comparison, first try to compare the whole string. If no match is
found, strip the parameters and try that way.

ASTERISK-29275 #close

Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f

Some code has been added referencing symbols defined in a block
protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
ifdef blocks too.

ASTERISK-29660

Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f

Adds parsing of ANI II digits (Originating
Line Information) to PJSIP, on par with
what currently exists in chan_sip.

ASTERISK-29472

Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847

Fixed the external media creation handle to handle the 'data' option correctly.

ASTERISK-29629

Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129

dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.

Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.

ASTERISK-29546

Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26