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.
cdr_mysql was deprecated in 1.8, to be removed in 19.
app_mysql was deprecated in 1.8, to be removed in 19.
app_ices was deprecated in 16, to be removed in 19.
app_macro was deprecated in 16, to be removed in 21.
app_fax was deprecated in 16, to be removed in 19.
app_url was deprecated in 16, to be removed in 19.
app_image was deprecated in 16, to be removed in 19.
app_nbscat was deprecated in 16, to be removed in 19.
app_dahdiras was deprecated in 16, to be removed in 19.
cdr_syslog was deprecated in 16, to be removed in 19.
chan_oss was deprecated in 16, to be removed in 19.
chan_phone was deprecated in 16, to be removed in 19.
chan_sip was deprecated in 17, to be removed in 21.
chan_nbs was deprecated in 16, to be removed in 19.
chan_misdn was deprecated in 16, to be removed in 19.
chan_vpb was deprecated in 16, to be removed in 19.
res_config_sqlite was deprecated in 16, to be removed in 19.
res_monitor was deprecated in 16, to be removed in 21.
conf2ael was deprecated in 16, to be removed in 19.
muted was deprecated in 16, to be removed in 19.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29554
ASTERISK-29555
ASTERISK-29557
ASTERISK-29558
ASTERISK-29559
ASTERISK-29560
ASTERISK-29561
ASTERISK-29562
ASTERISK-29563
ASTERISK-29564
ASTERISK-29565
ASTERISK-29566
ASTERISK-29567
ASTERISK-29568
ASTERISK-29569
ASTERISK-29570
ASTERISK-29571
ASTERISK-29572
ASTERISK-29573
ASTERISK-29574

Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

Adds function to selectively drop specified frames
in the TX or RX direction on a channel, including
control frames.

ASTERISK-29478

Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec

With Asterisk 1.6.0, in the main parser for the configuration file
extensions.conf, the separator was changed from vertical bar to comma.
However, the first separator was not changed in aelparse; it still had
to be a vertical bar, and no comma was allowed.

Additionally, this change allows the vertical bar for the first and
last parameter again, even in the main parser, because the vertical bar
was still accepted for the other parameters.

ASTERISK-29540

Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81

This format did not specify a "write" handler, so when attempting to write
to it (ast_writestream) a crash would occur.

This patch adds a default handler that simply issues a "not supported"
warning, thus no longer crashing.

ASTERISK-29539

Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91

Previously, if CDR filters were used so that
not all CDR records used all sections defined
in cdr_adaptive_odbc.conf, then warnings will
always be emitted (if each CDR record is unique
to a particular section, n-1 warnings to be
specific).

This turns the offending warning log into
a verbose message like the other one, since
this behavior is intentional and not
indicative of anything wrong.

ASTERISK-29494

Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8

Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.

ASTERISK-29528

Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a

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

If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
This makes the audio stream not-playable at the receiver side.
Linphone isn't able to play such an audio - lots of disruptions are heard.
Also I had complains of bad audio from users which use other types of phones.

After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).

Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).

However, this flag is never set in Asterisk-12 and newer.
Previously it has been set (see Asterisk-11).

ASTERISK-29526 #close

Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d

Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.

ASTERISK-29477

Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f

Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c

If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.

ASTERISK-29415 #close

Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.

This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.

ASTERISK-29392 #close

Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

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

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

Without this message, it is not obvious that the reason is STUN timeout.

ASTERISK-29507 #close

Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74

Use the URI parsing functions 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

This appears to just have been a copy/paste error from 6258bbe7. Fix
suggested by Ross Beer in ASTERISK~29166.

Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37

Adds an application to reload modules
from within the dialplan.

ASTERISK-29454

Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774

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

While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.

ASTERISK-29444

Change-Id: I08adf2824b8bc63405778cf355963b5005612f41

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

If the system time has stepped backwards because of a time
adjustment between the time a frame is timestamped and the
time we check the timestamps in abstract_jb:hook_event_cb(),
we get a negative interval, but we don't check for that there.
abstract_jb:hook_event_cb() then calls
fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
and the first thing that does is assert(interval >= 0).

There are several issues with this...

 * abstract_jb:hook_event_cb() saves the interval in a variable
   named "now" which is confusing in itself.

 * "now" is defined as an unsigned int which converts the negative
   value returned from ast_tvdiff_ms() to a large positive value.

 * fixed_jb_get()'s parameter is defined as a signed int so the
   interval gets converted back to a negative value.

 * fixed_jb_get()'s assert is NOT an ast_assert but a direct define
   that points to the system assert() so it triggers even in
   production mode.

So...

 * hook_event_cb()'s "now" was renamed to "relative_frame_start" and
   changed to an int64_t.
 * hook_event_cb() now checks for a negative value right after
   retrieving both the current and framedata timestamps and just
   returns the frame if the difference is negative.
 * fixed_jb_get()'s local define of ASSERT() was changed to call
   ast_assert() instead of the system assert().

ASTERISK-29480
Reported by: Dan Cropp

Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9

Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.

ASTERISK-29442

Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e

When using the Busy() and Congestion() applications the
function ast_safe_sleep is used by wait_for_hangup to safely
wait on the channel. This function may send silence if Asterisk
is configured to do so using the transmit_silence option.

In a scenario where an answered channel dials a Local channel
either directly or through call forwarding and the Busy()
or Congestion() dialplan applications were executed with the
transmit_silence option enabled the busy or congestion
tone would not be heard.

This is because inband generation of tones (such as busy
and congestion) is stopped when other audio is sent to
the channel they are being played to. In the given
scenario the transmit_silence option would result in
silence being sent to the channel, thus stopping the
inband generation.

This change adds a variant of ast_safe_sleep which can be
used when silence should not be played to the channel. The
wait_for_hangup function has been updated to use this
resulting in the tones being generated as expected.

ASTERISK-29485

Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133

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

Previously, SayNumber always emitted a warning if the caller hung up
during execution. Usually this isn't correct, so check if the channel
hung up and, if so, don't emit a warning.

ASTERISK-29475

Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594

The scenario where a channel still has an associated datastore we
cannot unload since there is a function pointer to the destroy and fixup
functions in play.  Thus increase the module ref count whenever we
allocate a datastore, and decrease it during destroy.

In order to tighten the race that still exists in spite of this (below)
add some extra failure cases to prevent allocations in these cases.

Race:

If module ref is zero, an LOCK or TRYLOCK is invoked (near)
simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
in such a case the datastore is created *prior* to unloading being set
to true (first step in module unload) then it's possible that the module
will unload with the destructor being called (and segfault) post the
module being unloaded.  The module will however wait for such locks to
release prior to unloading.

If post that we can recheck the module ref before returning the we can
(in theory, I think) eliminate the last of the race.  This race is
mostly theoretical in nature.

Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

For example:

arthur*CLI> dialplan locks show
func_lock locks:
Name                                     Requesters Owner
uls-autoref                              0          (unlocked)
1 total locks listed.

Obviously other potentially useful stats could be added (eg, how many
times there was contention, how many times it failed etc ... but that
would require keeping the stats and I'm not convinced that's worth the
effort.  This was useful to troubleshoot some other issues so submitting
it.

Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

AST_TRAVERSE accessess current as current = current->(field).next ...
and since we free current (and ast_free poisons the memory) we either
end up on a ast_mutex_lock to a non-existing lock that can never be
obtained, or a segfault.

Incidentally add logging in the "we have to wait for a lock to release"
case, and remove an ineffective statement that sets memory that was just
cleared by ast_calloc to zero.

Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

In two places we bail out with failure after we've already incremented
the requesters counter, if this occured then it would effectively result
in unload to wait indefinitely, thus preventing clean shutdown.

Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.

ASTERISK-29450

Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66