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

With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
-fPIC option added to its _ASTCFLAGS.

ASTERISK-29634

Change-Id: I34649c85e075fd954e578378fabf798c3f038f50

When compiled without extended srtp crypto suites also disable parsing
these from received SDP. This prevents using these, as some client
implementations are not stable.

ASTERISK-29625

Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a

Fixed ARI external media handler to accept body parameters.

ASTERISK-29622

Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d

This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.

The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.

ASTERISK-29508 #close

Change-Id: I7955a046293f913ba121bbd82153b04439e3465f

ASTERISK-29616

Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900

Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83

Allow mapping pjproject log messages to the Asterisk TRACE
log level.  The defaults were also changes to log pjproject
levels 3,4 to DEBUG and 5,6 to TRACE.  Previously 3,4,5,6
all went to DEBUG.

ASTERISK-29582

Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d

Commit 305ce3de added -Wno-parentheses-equality to Makefile.rules,
turning the previous two warning suppressions from commit e9520dbe
redundant. Let us remove the latter.

Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac

ASTERISK-29602

Change-Id: I6f0af0a959409cdbc6b185b1604301bafc872a5a

ASTERISK-29598

Change-Id: I8ef17023f55bf01f2e309b06f4778a8ca7252c91

ASTERISK-20339 #close

Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5

app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572

Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
It may be used to get all X- headers in case the actual set and names of headers unknown.

ASTERISK-29389

Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b

Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.

A flag has been introduced to allow meters to fallback to counters.


ASTERISK-29513

Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7

Asterisk first looks at the end of the URL to determine the file
extension of the returned audio, which in many cases will not work
because the URL may end with a query string or a URL fragment. If that
fails, Asterisk then looks at the Content-Type header and then finally
parses the URL to get the extension.

The order has been changed such that we look at the Content-Type
header first, followed by looking for the extension of the parsed
URL. We no longer look at the end of the URL, which was error prone.

ASTERISK-29527 #close

Change-Id: I1e3f83b339ef2b80661704717c23568536511032

If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.

ASTERISK-29381

Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042

Verify `ast_check_hangup` before looping to the next sound file.
If the call is already hangup we just break the cycle.
It also ensures that the PlaybackFinished event is sent if the call was hangup.

This is also use-full when we are playing a big list of file for a channel that is hangup.
Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.

With the patch we just break the playback cycle when the chan is hangup.

ASTERISK-29501 #close

Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8

ASTERISK-27871 #close

Change-Id: I6624f2d3a57f76a89bb372ef54a124929a0338d7

From RFC 8225 Section 5.2.1:

    The "dest" claim is a JSON object with the claim name of "dest"
    and MUST have at least one identity claim object.  The "dest"
    claim value is an array containing one or more identity claim JSON
    objects representing the destination identities of any type
    (currently "tn" or "uri").  If the "dest" claim value array
    contains both "tn" and "uri" claim names, the JSON object should
    list the "tn" array first and the "uri" array second.  Within the
    "tn" and "uri" arrays, the identity strings should be put in
    lexicographical order, including the scheme-specific portion of
    the URI characters.

Additionally, make it clear that there was a failure to sign the JWT
payload and not necessarily a memory allocation failure.

Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9

Use cURL's URL parsing API, falling back to the urlparser library, to
parse playback URLs in order to find their file extensions.

For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.

ASTERISK-27871 #close

Change-Id: I16d0682f6d794be96539261b3e48f237909139cb

Add check that data parameter specified when audiosocket used for externalMedia.

ASTERISK-29514 #close

Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617

In f8b0c2c9 we added support for port numbers in 'match' statements
but neglected to include that support in the PJSIP config wizard.

The removed code would have also prevented IPv6 addresses from being
successfully used in the config wizard as well.

ASTERISK-29503 #close

Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d

When we try to play a list of sound files in the same Play command,
we get only one PlaybackFinish event, after all sounds are played.

But in the case where the Play fails (because channel is destroyed
for example), Asterisk will send one PlaybackFinish event for each
sound file still to be played. If the list is big, Asterisk is
sending many events.

This patch adds a failed state so we can understand that the play
failed. On that case we don't send the event, if we still have a
list of sounds to be played.

When we reach the last sound, we send the PlaybackFinish with
the failed state.

ASTERISK-29464 #close

Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322

With the fix for ASTERISK_28754 channels are no longer put on hold if an
outbound INVITE is answered with a "Session Progress" containing
"inactive" audio.

The previous change moved the evaluation of the media attributes to
`negotiate_incoming_sdp_stream()` to have the `remotely_held` status
available when building the SDP in `create_outgoing_sdp_stream()`.
This however means that an answer to an outbound INVITE, which does not
traverse `negotiate_incoming_sdp_stream()`, cannot set the
`remotely_held` status anymore.

This change moves the check so that both, `negotiate_incoming_sdp_stream()` and
`apply_negotiated_sdp_stream()` can do the checks.

ASTERISK-29479

Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75

When the MessageSend destination is in the form
PJSIP/<number>@<endpoint> and the endpoint's contact
URI already has a user component, that user component
will now be replaced with <number> when creating the
request URI.

ASTERISK_29404

Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5

Set preferred transport when querying the local address to use in
filter_on_tx_messages(). This prevents the module to erroneously select
the wrong transport if more than one transports of the same type (TCP or
TLS) are configured.

ASTERISK-29241

Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6

The text description needs to be the last thing on the AST_MODULE_INFO
line to be pulled in properly by menuselect.

Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832

Adds hook flash recognition support
for application/hook-flash.

ASTERISK-29460

Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea

 * Implemented the new "to" parameter of the MessageSend()
   dialplan application.  This allows a user to specify
   a complete SIP "To" header separate from the Request URI.

 * Completely refactored the get_outbound_endpoint() function
   to actually handle all the destination combinations that
   we advertized as supporting.

 * We now also accept a destination in the same format
   as Dial()...  PJSIP/number@endpoint

 * Added lots of debugging.

ASTERISK-29404
Reported by Brian J. Murrell

Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce

STIR/SHAKEN requires a Date header alongside the Identity header, so
that has been added. Still on the outgoing side, we were missing the
dest->tn section of the JSON payload, so that has been added as well.
Moving to the incoming side, URL checking has been added to the public
cert URL to ensure that it starts with http.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c

For connection oriented transports PJSIP uses factories to
produce transports. When doing a partial transport reload
we need to also move the factory of the transport over so
that anything referencing the transport (such as an endpoint)
has the factory available.

ASTERISK-29441

Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161

When unsubscribing from an endpoint technology a FRACK
would occur due to incorrect reference counting. This fixes
that issue, along with some other issues.

Fixed a typo in get_subscription when calling ao2_find as it
needed to pass the endpoint ID and not the entire object.

Fixed scenario where a subscription would get returned when
it shouldn't have been when searching based on endpoint
technology.

A doulbe unreference has also been resolved by only explicitly
releasing the reference held by tech_subscriptions.

ASTERISK-28237 #close
Reported by: Lucas Tardioli Silveira

Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729

In multidomain environments, it is desirable to create
PJSIP endpoints with the domain info in the endpoint name
in pjsip_endpoint.conf.  This resulted in an error with
registrations, NOTIFY, and OPTIONS packet generation.

This commit will detect if there is an @ in the endpoint
identifier and generate the URI accordingly so NOTIFY and
OPTIONS From headers will generate correctly.

ASTERISK-28393

Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619

RTCP ICE candidates use a base address derived from the RTP
candidate. The port on the base address was not being updated to
the RTCP port.

This change sets the base port to the RTCP port and all is well.

ASTERISK-29433

Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040

By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:

https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2

This patch adds a `stun_software_attribute` yes/no option to make it
possible to omit the SOFTWARE attribute from STUN packets.

ASTERISK-29434

Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b