If a switch is invoked using chan_iax2, deadlock can result
because the PBX core is autoservicing the channel while chan_iax2
also then attempts to service it while waiting for the result
of the switch. This removes servicing of the channel to prevent
any conflicts.

ASTERISK-30064 #close

Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22

If tab completion using ast_module_helper is attempted
during startup, deadlock will ensue because the CLI
will attempt to lock the module list while it is already
locked by the loader. This causes deadlock because when
the loader tries to acquire the CLI lock, they are blocked
on each other.

Waiting for startup to complete is not feasible because
the CLI lock is acquired while waiting, so deadlock will
ensure regardless of whether or not a lock on the module
list is attempted.

To prevent deadlock, we immediately abort if tab completion
is attempted on the module list before Asterisk is fully
booted.

ASTERISK-30039 #close

Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d

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

When a PJSIP channel is set on hold or off hold, all streams were set
on/off hold. This is not the desired behaviour and caused issues
when there were multiple streams in the topology.

Now, only the default audio stream is set on/off hold when a hold is
indicated.

ASTERISK-30051

Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71

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

Make sure that we have a working sed before trying to use it.

ASTERISK-30059 #close

Change-Id: I9abad67a5df11b665d480feec304ab9d6f55cc76

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

ASTERISK-30060 #close

Change-Id: I88d47a1488be2f39017b8d562f993f081844fcb8

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

Adds the CONFBRIDGE_CHANNELS function which can be used
to retrieve a comma-separated list of channels, filtered
by a particular type of participant category. This output
can then be used with functions like UNSHIFT, SHIFT, POP,
etc.

ASTERISK-30036 #close

Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83

Currently, the operator services mode in DAHDI is broken and unusable.
The actual operator recall functionality works properly; however,
when the operator hangs up (which is the only way that such a call
is allowed to end), both lines are permanently taken out of service
until "dahdi restart" is run. This prevents this feature from being
used.

Operator mode is one of the few factors that can cause the general
analog event handling in sig_analog not to be used. Several years
back, much of the analog handling was moved from chan_dahdi to
sig_analog. However, this was not done fully or consistently at
the time, and when operator mode is active, sig_analog does not
get used. Generally this is correct, but in the case of hangup
it should be using sig_analog regardless of the operator mode;
otherwise, the lines do not properly clear and they become unusable.

This bug is fixed so the operator can now hang up and properly
release the call. It is treated just like any other hangup. The
operator mode functionality continues to work as it did before.

ASTERISK-29993 #close

Change-Id: Ib2e3ddb40d9c71e8801e0b4bb0a12e2b52f51d24

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

GCC 12 caught an issue in state_id_by_topic where we were
checking a pointer for NULL instead of the contents of
the pointer for '\0'.

ASTERISK-30044

Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e

When a new unreal (local) channel is created, a second (;2) channel is
created as a counterpart which clones the topology of the first
channel. This creates issues when an outgoing stream is sendonly or
recvonly as the stream state of the inbound channel will be the same
as the stream state of the outbound channel.

Now the stream state is flipped for the streams of the 2nd channel in
ast_unreal_new_channels if the outgoing stream topology is recvonly or
sendonly.

ASTERISK-29655
Reported by: Michael Auracher

ASTERISK-29638
Reported by: Michael Auracher

Change-Id: I0cea29635bb20b7bf7fd0fb95498cd44dab98fbf

Documents the Dial syntax for DAHDI, namely the channel group,
distinctive ring, answer confirmation, and digital call options
that are specified in the resource itself.

ASTERISK-24827 #close

Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30

For lines that have mailboxes configured on them, with
FSK MWI, DAHDI will periodically try to dispatch FSK
to update MWI. However, this is never supposed to be
done when a channel is not idle.

There is currently an edge case where MWI FSK can
extraneously get spooled for the channel if a caller
hook flashes and hangs up, which triggers a recall ring.
After one ring, the on hook time threshold in this if
condition has been satisfied and an MWI update is spooled.
This means that when the phone is picked up again, the
answerer gets an FSK spill before being reconnected to
the party on hold.

To prevent this, we now explicitly check to ensure that
subchannel 0 has no owner. There is no owner when DAHDI
channels are idle, but if the channel is "in use" in some
way (such as in the aforementioned scenario), then there
is an owner, and we shouldn't process MWI at this time.

ASTERISK-28518 #close

Change-Id: Ia3904434fd81688d71742f7e84358b7e1c38e92a

Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.

ASTERISK-29931
Added by Michael Cargile

Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2

Currently, if any custom ring cadences are specified, they are
appended to the array of cadences from wherever we left off
last time. This works properly the first time, but on subsequent
dahdi restarts, it means that the existing cadences are left
alone and (most likely) the same cadences are then re-added
afterwards. In short order, the cadence array gets maxed out
and the user begins seeing warnings that the array is full
and no more cadences may be added.

This buggy behavior persists until Asterisk is completely
restarted; however, if and when dahdi restart is run again,
then the same problem is reintroduced.

This fixes this behavior so that cadence parsing is more
idempotent, that is so running dahdi restart multiple times
starts adding cadences from the beginning, rather than from
wherever the last cadence was added.

As before, it is still not possible to revert to the default
cadences by simply removing all cadences in this manner, nor
is it possible to delete existing cadences. However, this
does make it possible to update existing cadences, which
was not possible before, and also ensures that the cadences
remain unchanged if the config remains unchanged.

ASTERISK-29990 #close

Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d

Currently, if attempting to place a call to a peer that only allows
RSA authentication, if we fail to provide an outkey when placing
the call, Asterisk will crash.

This exposes the broader issue that IAX2 is prone to causing a crash
if encryption or decryption is attempted but we never initialized
the encryption and decryption keys. In other words, if the logic
to use encryption in chan_iax2 is not perfectly aligned with the
decision to build keys in the first place, then a crash is not
only possible but probable. This was demonstrated by ASTERISK_29264,
for instance.

This permanently prevents such events from causing a crash by explicitly
checking that keys are initialized properly before setting the flags
to use encryption for the call. Instead of crashing, the call will
now abort.

ASTERISK-30007 #close

Change-Id: If925c3d86099ceac7f621804f2532baac5050c9a

A bug in menuselect can cause modules that are disabled
by default to be recompiled every time a recompilation
occurs. This occurs for module categories that are NOT
positive output, as for these categories, the modules
contained in the makeopts file indicate modules which
should NOT be selected. The existing procedure of iterating
through these modules to mark modules as present is thus
insufficient. This has led to modules with a default_enabled
tag of "no" to get deleted and recompiled every time, even
when they haven't changed.

To fix this, we now modify the mark as present behavior
for module categories that are not positive output. For
these, we start by iterating through the module tree
and marking all modules as present, then go back and
mark anything contained in the makeopts file as not
present. This ensures that makeopt selections are actually
used properly, regardless of whether a module category
uses positive output or not.

ASTERISK-29728 #close

Change-Id: Idf2974c4ed8d0ba3738a92f08a6082b234277b95

The admin_exec function in app_meetme is used by the SLA
applications for internal bridging. However, in these cases,
chan is NULL. Currently, this function will set some status
variables that are intended for a channel, but since channel
is NULL, this is erroneously creating meaningless global
variables, which shouldn't be happening. This sets these
variables only if chan is not NULL.

ASTERISK-30002 #close

Change-Id: I817df6c26f5bda131678e56791b0b61ba64fc6f7

Some command line options to Asterisk only apply when Asterisk
is started and cannot be used with remote console mode. If a
user tries to use any of these, they are currently simply
silently ignored.

This prints out a warning if incompatible options are used,
informing users that an option used cannot be used with remote
console mode. Additionally, some clarifications are added to
the help text and man page.

ASTERISK-22246
ASTERISK-26582

Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337

Adds the DB_KEYCOUNT function, which can be used to retrieve
the number of keys at a given prefix in AstDB.

ASTERISK-29968 #close

Change-Id: Ib2393b77b7e962dbaae6192f8576bc3f6ba92d09

According to chan_dahdi.conf, up to 64 groups (numbered
0 through 63) can be used when dialing DAHDI channels.

However, currently dialing round robin with a group number
greater than 31 fails because the array for the round robin
structure is only size 32, instead of 64 as it should be.

This fixes that so the round robin array size is consistent
with the actual groups capacity.

ASTERISK-29994

Change-Id: I4caa08d7025f78ac75a0539f71aaf3eb3e85b3b7

If Asterisk receives a SIP REFER with Session-Timers UAC
maintain Session-Timers when sending UPDATE"

ASTERISK-29843

Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c

This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.

ASTERISK-29486

Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432

Currently, if a user uses an application like ControlPlayback
to try to rewind a file past the beginning, this can throw
warnings when the file format (e.g. PCM) tries to seek to
a negative offset.

Instead of letting file formats try (and fail) to seek a
negative offset, we instead now catch this in the rewind
function to ensure that we never seek an offset less than 0.
This prevents legitimate user actions from triggering warnings
from any particular file formats.

ASTERISK-29943 #close

Change-Id: Ia53f2623f57898f4b8e5c894b968b01e95426967

PJSIP currently is capable of receiving flash events
and converting them to FLASH control frames, but it
currently lacks support for doing the reverse: taking
a FLASH control frame and converting it into a flash
event in the SIP domain.

This adds the ability for PJSIP to process flash control
frames by converting them into the appropriate SIP INFO
message, which can then be sent to the peer. This allows,
for example, flash events to be sent between Asterisk
systems using PJSIP.

ASTERISK-29941 #close

Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd

Adds the dialplan eval function commands to evaluate a dialplan
function from the CLI. The return value and function result are
printed out and can be used for testing or debugging.

ASTERISK-29820 #close

Change-Id: I833e97ea54c49336aca145330a2adeebfad05209

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

Removes a couple sample config files for modules
which have since been removed from Asterisk.

ASTERISK-30008 #close

Change-Id: I6be89cafc6c575d98a5315e4912b61a833aacf52

added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183

ASTERISK-29842

Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18

if Asterisk need to send an UPDATE before answer
on a channel that uses Record-Route:
it will not include a Route header

ASTERISK-29955

Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef

On a write error to an AMI session a flag was set to
indicate that the write error had occurred, with the
expected result being that the session be terminated.
This was not actually happening and instead writing
would continue to be attempted.

This change adds a check for the write error and causes
the session to actually terminate.

ASTERISK-29948

Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5

Patch provided inline by Yury Kirsanov on the linked issue and
approved by Josh Colp.

ASTERISK-29253 #close

Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f

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