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

When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.

Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.

ASTERISK-29655
Reported by: Michael Auracher

ASTERISK-29638
Reported by: Michael Auracher

Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1

There was an issue with the conditional where STIR/SHAKEN would be
enabled even when not configured. It has been changed to ensure that if
a profile does not exist and stir_shaken is not set in pjsip.conf, then
the conditional will return from the function without performing
STIR/SHAKEN operations.

ASTERISK-30024

Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6

The async_operations setting on a transport configures how
many simultaneous incoming packets the transport can handle
when multiple threads are polling and waiting on the transport.
As we only use a single thread this was needlessly creating
incoming packets when set to a non-default value, wasting memory.

ASTERISK-30006

Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f

ASTERISK-30021 #close

Change-Id: I70eb59b782a4946b979942e21422746b7563029c

ASTERISK-30023

Change-Id: I0e1697f6af044e9eab7e07bbaeeffd1bb68ac34a

Chrome has added more attributes, causing the limit to be
exceeded. This raises it up some more.

ASTERISK-30015

Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32

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

Some databases depending on their configuration using backslashes
for escaping. When combined with the use of ' this can result in
a broken func_odbc query.

This change adds a SQL_ESC_BACKSLASHES dialplan function which can
be used to escape the backslashes.

This is done as a dialplan function instead of being always done
as some databases do not require this, and always doing it would
result in incorrect data being put into the database.

ASTERISK-29838

Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d

The ReceiveMF and ReceiveSF applications currently always
return 0, even if a channel has hung up. The call will still
end but generally applications are expected to return -1 if
the channel has hung up.

We now return -1 if a hangup occured to bring this behavior
in line with this norm. This has no functional impact, but
merely increases conformity with how these modules interact
with the PBX core.

ASTERISK-29951 #close

Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033

Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.

This option functions like the m option to Dial.

ASTERISK-29876 #close

Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7

Currently, if a user tries to access a non-dynamic
MeetMe conference and the conference is not found,
the call simply silent hangs up. There is no indication
to the user that anything went wrong at all.

This changes the relevant debug message to a warning
so that the user is notified of this invalidity.

ASTERISK-29954 #close

Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048

Removes some leftover build and config references to
modules that have since been removed from Asterisk.

ASTERISK-29935 #close

Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af

When adding headers to an outgoing request the headers were cloned using
the dialog's pool when they should have been cloned using tdata's pool.
Under certain circumstances it was possible for the dialog object, and
its pool to be freed while tdata is still active and available. Thus the
cloned header "disappeared", and when tdata tried to later access it a
crash would occur.

This patch makes it so all added headers are cloned appropriately using
tdata's pool.

ASTERISK-29411 #close
ASTERISK-29535 #close

Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5

Several modules removal and deprecations occurred in 19.0.0 (initial
19 release), but associated UPGRADE files were not removed from
staging for some reason in the master branch.

This patch removes those files, and also removes a spurious leftover
header, chan_phone.h (associated module removed in 19).

Change-Id: Ib92142c846b45c882d6b2b6caca7225253c83add

This change removes patches which have been merged into
upstream and updates some existing ones. It also adds
some additional config_site.h changes to restore previous
behavior, as well as a patch to allow multiple Authorization
headers. There seems to be some confusion or disagreement
on language in RFC 8760 in regards to whether multiple
Authorization headers are supported. The RFC implies it
is allowed, as does some past sipcore discussion. There is
also the catch all of "local policy" to allow it. In
the case of Asterisk we allow it.

ASTERISK-29351

Change-Id: Id39ece02dedb7b9f739e0e37ea47d76854af7191

The PBX core uses the stack when it comes to includes, which
means that a context can only contain strictly fewer than
AST_PBX_MAX_STACK includes. If this is exceeded, then warnings
will be emitted for each number of includes beyond this if
searching for an extension in the including context, and if
the extension's inclusion is beyond the stack size, it will
simply not be found.

To address this, we now check if there are too many includes
in a context when the dialplan is reloaded so that if there
is an issue, the user is aware of at "compile time" as opposed
to "run time" only. Secondly, more details are printed out
when this message is encountered so it's clear what has happened.

ASTERISK-26719

Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958

make_xml_documentation was being called with the --validate
flag set when it shouldn't have been.  This was causing
build failures if neither xmllint nor xmlstarlet were installed.
The correct behavior is to simply print a message that either
one of those tools should be installed for validation and
continue with the build.

ASTERISK-29988

Change-Id: Idc6c44114e7dd3fadae183a4e22f4fdba0b8a645

get_sourceable_makeopts wasn't handling variables with embedded
double quotes in them very well.  One example was the DOWNLOAD
variable when curl was being used instead of wget.  Rather than
trying to fix get_sourceable_makeopts, it's just been removed.

ASTERISK-29986
Reported by: Stefan Ruijsenaars

Change-Id: Idf2a90902228c2558daa5be7a4f8327556099cd2

The iax2 show netstats command previously didn't contain
enough spacing in the header to properly align the table
header with the table body. This caused column headers
to not align with the values on longer channel names.

Some spacing is added to account for the longest channel
names that display (before truncation occurs) so that
columns are always properly aligned.

ASTERISK-29895 #close
patches:
  61205_misaligned2.patch submitted by Birger Harzenetter (license 5870)

Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f

There is work going on to update our OpenSSL usage to avoid the
deprecated functions but in the meantime make it possible to compile
in devmode.

Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726

Adding information in the readme about running the install_preqreq script to install components that the ./configure script might indicate as missing.

ASTERISK-29976 #close

Change-Id: Ic287b46300168729838bddd8f9265e98fc22bce6

ASTERISK_22025 introduced a regression that shows
the host IP and port as the perceived IP and port
again, as opposed to showing the actual perceived
address. This fixes this by showing the correct
information.

ASTERISK-29048 #close

Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a

Change RTP timer behavior for detecting RTP only after two-way
SDP channel establishment. Ignore detecting after receiving 183
with SDP or while direct media is used.
Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
and rtpholdtimeout options in chan_sip.

ASTERISK-26689 #close
ASTERISK-29929 #close

Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb