Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.

ASTERISK-29529

Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702

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

Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.

This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.

Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.

ASTERISK-29531

Change-Id: If9718c89353b8e153d84add3cc4637b79585db19

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

IPv6 nameserver addresses are stored in different part of the
__res_state structure, so look there if we appear to have support for
it.

ASTERISK-28004 #close

Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d

Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.

ASTERISK-18454 #close

Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b

ASTERISK-29598

Change-Id: I8ef17023f55bf01f2e309b06f4778a8ca7252c91

ASTERISK-29597

Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1

ASTERISK-29596

Change-Id: Ibae9490c1b35cadbf7028d24610f745277c8535e

ASTERISK-29595

Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33

ASTERISK-29593

Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99

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

Bumped AMI and ARI versions for the next major Asterisk version (20).

Change-Id: I2e65794f206d443178ab6895767fb53f04cc3e6a

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

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

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

RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm.  The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred.  We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.

The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.

It doesn't stop there though... Each realm can require a
different username from the others.  There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.

So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header.  We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.

In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.

NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5.  We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5.  When we support the
others, we'll move the check into the loop that searches the
objects.

Changes:

 * Added a new API ast_sip_retrieve_auths_vector() that takes in
   a vector of auth ids (usually supplied on a call to
   ast_sip_create_request_with_auth()) and populates another
   vector with the actual objects.

 * Refactored res_pjsip_outbound_authenticator_digest to handle
   multiple Authenticate headers and set the stage for handling
   additional digest algorithms.

 * Added a pjproject patch that allows them to ignore digest
   algorithms they don't support.  This patch has already been
   merged upstream.

 * Updated documentation for auth objects in the XML and
   in pjsip.conf.sample.

 * Although res_pjsip_authenticator_digest isn't affected
   by this change, some debugging and a testsuite AMI event
   was added to facilitate testing.

Discovered during OpenSIPit 2021.

ASTERISK-29397

Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281

Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.

This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.

ASTERISK-29380

Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81

STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.

The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.

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

Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c

During OpenSIPit, we found out that the public certificates must be of
type X.509. When reading in public keys, we use the corresponding X.509
functions now.

We also discovered that we needed a better naming scheme for the
certificates since certificates with the same name would cause issues
(overwriting certs, etc.). Now when we download a public certificate, we
get the serial number from it and use that as the name of the cached
certificate.

The configuration option public_key_url in stir_shaken.conf has also
been renamed to public_cert_url, which better describes what the option
is for.

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

Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d

ASTERISK-27477 #close

Change-Id: I68f6715bba92a525149e35d142a49377a34a1193

Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.

Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392

The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.

Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
  main/logger.c: static char dateformat[256] = "%b %e %T"

This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages

Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko

Change-Id: I5c104dc1f8417ccd3d01faf86e84ccbf89bc3b31
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Because they modify their argument they are not pure functions and
should not be marked as such, otherwise the compiler may optimize
them away.

ASTERISK-29306 #close

Change-Id: Ibec03a08522dd39e8a137ece9bc6a3059dfaad5f

Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.

This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.

ASTERISK-29321

Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1

This partially reverts commit 3d1bf3c5,
specifically for app.h.

This works with both gcc 9.3.0 and 10.2.0 now, both for C and C++ (as
tested with external modules).

ASTERISK-29287

Change-Id: I5b9f02a9b290675682a1d13f1788fdda597c9fca
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

ASTERISK-29244

Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5

After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.

ASTERISK-29035 #close

Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

Rename check_manager_enabled() and check_webmanager_enabled() to begin
with ast_ so that the symbols are automatically exported by the
linker.

ASTERISK~29184

Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9

This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.

Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940

As described in the issue, /tmp is not a suitable location for a
large amount of cached media files, since most distributions make
/tmp a RAM-based tmpfs mount with limited capacity.

I opted for a location that can be configured separately, as opposed
to using a subdirectory of spooldir, given the different storage
profile (transient files vs files that might stay there indefinitely).

This commit just makes the cache directory configurable, and changes
the default location from /tmp to /var/cache/asterisk.

ASTERISK-29143

Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d

* Added a ONESHOT type that never reschedules.

* Added "like" capability to "pjsip show scheduled_tasks" so you can do
  the following:

  CLI> pjsip show scheduled_tasks like outreg
  PJSIP Scheduled Tasks:

  Task Name                                     Interval  Times Run ...
  ============================================= ========= ========= ...
  pjsip/outreg/testtrunk-reg-0-00000074            50.000   oneshot ...
  pjsip/outreg/voipms-reg-0-00000073              110.000   oneshot ...

* Fixed incorrect display of "Next Start".

* Compacted the displays of times in the CLI.

* Added two new functions (ast_sip_sched_task_get_times2,
  ast_sip_sched_task_get_times_by_name2) that retrieve the interval,
  next start time, and next run time in addition to the times already
  returned by ast_sip_sched_task_get_times().

Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3

The data can be freed if the old object '_data' is the same object as
new 'data'. Because at first the object is unreferenced which can lead
to destroying it.

This could happened in res_pjsip_pubsub when the publication is updated
which could lead to segfault in function publish_expire.

Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da

This avoids the inclusion of the OpenSSL headers in the public header,
which avoids one external library dependency in res_pjsip_stir_shaken.

Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4

pjproject returns the dialog locked and with a reference. However,
in Asterisk the method that handles this decrements the reference
and removes the lock prior to returning. This makes it possible,
under some circumstances, for another thread to free said dialog
before the thread that created it attempts to use it again. Of
course when the thread that created it tries to use a freed dialog
a crash can occur.

This patch makes it so Asterisk now returns the newly created
dialog both locked, and with an added reference. This allows the
caller to de-reference, and unlock the dialog when it is safe to
do so.

In the case of a new SIP Invite the lock, and reference are now
held for the entirety of the new invite handling process.
Otherwise it's possible for the dialog, or its dependent objects,
like the transaction, to disappear. For example if there is a TCP
transport error.

ASTERISK-29057 #close

Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
(cherry picked from commit 6baa4b53)

If Asterisk sends out and INVITE and receives a challenge with a
different nonce value each time, it will continually send out INVITEs,
even if the call is hung up. The endpoint must be configured for
outbound authentication in order for this to occur. A limit has been set
on outbound INVITEs so that, once reached, Asterisk will stop sending
INVITEs and the transaction will terminate.

ASTERISK-29013

Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7

Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:

  dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
  stun, stun_packet

These debug categories can be enable/disable via an Asterisk CLI command.

While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).

ASTERISK-29054 #close

Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849

In the event that the desired extension already exists,
ast_add_extension2_lockopt() will free the 'data' it is passed before
returning an error, so we should not be freeing it ourselves.

Additionally, there were two places where ast_add_extension2_lockopt()
could return an error without also freeing the 'data' pointer, so we
add that.

ASTERISK-29097 #close

Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae