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

ASTERISK-29326

Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a

A frame suppression API exists as part of channels
which allows audio frames to or from a channel to
be dropped. The MuteAudio AMI action uses this
API to perform its job.

This API uses a framehook to intercept flowing
audio and drop it when appropriate. It is the
responsibility of the framehook to free the
frame it is given if it changes the frame. The
suppression API failed to do this resulting in
a leak of audio frames.

This change adds the freeing of these frames.

ASTERISK-29071

Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b

Implemented the english way of saying the year in ast_say_date_with_format_nl.
Currently the numbers are spoken correctly until 2020 and stopped working
this year.

ASTERISK-29297 #close
Reported-by: Jacek Konieczny

Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4

Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.

This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.

Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34

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

ASTERISK-29259

Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5

Log proper control frame names instead of "Unknown control '14'", etc.

Change-Id: I1724f2f4d1b064b25a5c93a7da0cb03be5143935

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

The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.

    https://wiki.asterisk.org/wiki/x/Xc5uAg

However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.

For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.

ASTERISK-28883 #close

Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f

Scope tracing allows you to not specify a format string or variable,
in which case it just prints the indent, file, function, and line
number.  The trace output automatically adds a newline to the end
in this case.  If you also have debugging turned on for the module,
a debug message is also printed but the standard log functionality
which prints it doesn't add the newline so you have messages
that don't break correctly.

 * format_log_message_ap(), which is the common log
   message formatter for all channels, now adds a
   newline to the end of format strings that don't
   already have a newline.

ASTERISK-29209
Reported by: Alexander Traud

Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da

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

Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6

Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641.
common_recall_channel_setup() setups common things on the recalled transfer
target, but used same target as source instead trasfered.

ASTERISK-29161 #close

Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f

ASTERISK-29146

Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41

Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0
Warning:
  say.c:2371:24: error: ‘%d’ directive output may be truncated writing
    between 1 and 11 bytes into a region of size 10
    [-Werror=format-truncation=]
  2371 |     snprintf(buf, 10, "%d", num);
  say.c:2371:23: note: directive argument in the range [-2147483648, 9]

That's not possible though, as the if() starts out checking for (num < 0),
making this Warning a false positive.

(Also replaced some else<TAB>if with else<SP>if while in the vicinity.)

Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a

ASTERISK-28430 #close

Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606

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

app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge.  To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second.  The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".

Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a

Check result of ast_translator_build_path against NULL before dereferencing.

ASTERISK-29091

Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29

ASTERISK-28311 #close

Change-Id: Ib1ce8fc1a8752751f5bf3615c59245532dfd9aa2

ASTERISK-28416 #close

Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4

Added to:
 * bridges/bridge_softmix.c
 * channels/chan_pjsip.c
 * include/asterisk/res_pjsip_session.h
 * main/channel.c
 * res/res_pjsip_session.c

There NO functional changes in this commit.

Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3

When both Asterisk and a UA send re-invites at the same time, both
send 491 "Transaction in progress" responses to each other and back
off a specified amount of time before retrying. When Asterisk
prepares to send its re-invite, it sets up the session's pending
media state with the new topology it wants, then sends the
re-invite.  Unfortunately, when it received the re-invite from the
UA, it partially processed the media in the re-invite and reset
the pending media state before sending the 491 losing the state it
set in its own re-invite.

Asterisk also was not tracking re-invites received while an existing
re-invite was queued resulting in sending stale SDP with missing
or duplicated streams, or no re-invite at all because we erroneously
determined that a re-invite wasn't needed.

There was also an issue in bridge_softmix where we were using a stream
from the wrong topology to determine if a stream was added.  This also
caused us to erroneously determine that a re-invite wasn't needed.

Regardless of how the delayed re-invite was triggered, we need to
reconcile the topology that was active at the time the delayed
request was queued, the pending topology of the queued request,
and the topology currently active on the session.  To do this we
need a topology resolver AND we need to make stream named unique
so we can accurately tell what a stream has been added or removed
and if we can re-use a slot in the topology.

Summary of changes:

 * bridge_softmix:
   * We no longer reset the stream name to "removed" in
     remove_all_original_streams().  That was causing  multiple streams
     to have the same name and wrecked the checks for duplicate streams.

   * softmix_bridge_stream_sources_update() was checking the old_stream
     to see if it had the softmix prefix and not considering the stream
     as "new" if it did.  If the stream in that slot has something in it
     because another re-invite happened, then that slot in old might
     have a softmix stream but the same stream in new might actually
     be a new one.  Now we check the new_stream's name instead of
     the old_stream's.

 * stream:
   * Instead of using plain media type name ("audio", "video", etc) as
     the default stream name, we now append the stream position to it
     to make it unique.  We need to do this so we can distinguish multiple
     streams of the same type from each other.

   * When we set a stream's state to REMOVED, we no longer reset its
     name to "removed" or destroy its metadata.  Again, we need to
     do this so we can distinguish multiple streams of the same
     type from each other.

 * res_pjsip_session:
   * Added resolve_refresh_media_states() that takes in 3 media states
     and creates an up-to-date pending media state that includes the changes
     that might have happened while a delayed session refresh was in the
     delayed queue.

   * Added is_media_state_valid() that checks the consistency of
     a media state and returns a true/false value. A valid state has:
     * The same number of stream entries as media session entries.
         Some media session entries can be NULL however.
     * No duplicate streams.
     * A valid stream for each non-NULL media session.
     * A stream that matches each media session's stream_num
       and media type.

   * Updated handle_incoming_sdp() to set the stream name to include the
     stream position number in the name to make it unique.

   * Updated the ast_sip_session_delayed_request structure to include both
     the pending and active media states and updated the associated delay
     functions to process them.

   * Updated sip_session_refresh() to accept both the pending and active
     media states that were in effect when the request was originally queued
     and to pass them on should the request need to be delayed again.

   * Updated sip_session_refresh() to call resolve_refresh_media_states()
     and substitute its results for the pending state passed in.

   * Updated sip_session_refresh() with additional debugging.

   * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE
     to pjproject if a transaction is in progress.  This stops us from
     creating a partial pending media state that would be invalid later on.

   * Updated reschedule_reinvite() to clone both the current pending and
     active media states and pass them to delay_request() so the resolver
     can tell what the original intention of the re-invite was.

   * Added a large unit test for the resolver.

ASTERISK-29014

Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb

Currently, it was not possible to create bridge with video_mode single.
This made hard to put the bridge in a vidoe_single mode.
So, added video_single option for Bridge creation using the ARI.
This allows create a bridge with video_mode single.

ASTERISK-29055

Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae

There's a race condition with bridging where a bridge can be torn down
causing the bridge_channel's ast_channel to become NULL when it's still
needed. This particular case happened with attended transfers, but the
crash occurred when trying to publish a stasis message. Now, the
bridge_channel is locked, a ref to the ast_channel is obtained, and that
ref is passed down the chain.

Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814

Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9

When the ExtensionState AMI action is executed on a pattern matched
hint it can end up adding a new hint if one does not already exist.
This results in a locking order of contexts -> hints -> contexts.

If at the same time a reload is occurring and adding its own hint
it will have a locking order of hints -> contexts.

This results in a deadlock as one thread wants a lock on contexts
that the other has, and the other thread wants a lock on hints
that the other has.

This change enforces a hints -> contexts locking order by explicitly
locking hints in the places where a hint is added when queried for.
This matches the order seen through normal adding of hints.

ASTERISK-29046

Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504

Added a new log formatter called "plain" that always prints
file, function and line number if available (even for verbose
messages) and never prints color control characters.  It also
doesn't apply any special formatting for verbose messages.
Most suitable for file output but can be used for other channels
as well.

You use it in logger.conf like so:
debug => [plain]debug
console => [plain]error,warning,debug,notice,pjsip_history
messages => [plain]warning,error,verbose

Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d

T.140 data in RTP is not zero terminated, so when we are queuing a text
frame on a bridge we need to ensure that we are passing a zero
terminated string.

ASTERISK-28974 #close

Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3

The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
at the same level as the scope level.  This allows the same
messages to be printed to the debug log when AST_DEVMODE
isn't enabled.

Also added a few variants of the SCOPE_EXIT macros that will
also call ast_log instead of ast_debug to make it easier to
use scope tracing and still print error messages.

Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21

 * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros
   which are shortcuts for
      ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP))

 * Added the stream position to the string representation of the
   stream.

 * Fixed some formatting in ast_stream_to_str().

Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b

Allow passing a topology from the called channel back to the
calling channel.

 * Added a new function ast_queue_answer() that accepts a stream
   topology and queues an ANSWER CONTROL frame with it as the
   data.  This allows the called channel to indicate its resolved
   topology.

 * Added a new virtual function to the channel tech structure
   answer_with_stream_topology() that allows the calling channel
   to receive the called channel's topology.  Added
   ast_raw_answer_with_stream_topology() that invokes that virtual
   function.

 * Modified app_dial.c and features.c to grab the topology from the
   ANSWER frame queued by the answering channel and send it to
   the calling channel with ast_raw_answer_with_stream_topology().

 * Modified frame.c to automatically cleanup the reference
   to the topology on ANSWER frames.

Added a few debugging messages to stream.c.

Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c

With the addition of STIR/SHAKEN, the function ast_base64decode_string
was added for convenience since there is a lot of converting done during
the STIR/SHAKEN process. This function returned the decoded string for
you, but did not NULL terminate it, causing some issues (specifically
with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the
documentation has been updated to reflect this.

Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5

There are various places in Asterisk - specifically in regards to
database integration - where having some kind of UTF-8 validation would
be beneficial. This patch adds:

* Functions to validate that a given string contains only valid UTF-8
  sequences.

* A function to copy a string (similar to ast_copy_string) stopping when
  an invalid UTF-8 sequence is encountered.

* A UTF-8 validator that allows for progressive validation.

All of this is based on the excellent UTF-8 decoder by Björn Höhrmann.
More information is available here:

    https://bjoern.hoehrmann.de/utf-8/decoder/dfa/

The API was written in such a way that should allow us to replace the
implementation later should we determine that we need something more
comprehensive.

Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9

Currently, if the bridge has created by the ARI, the video_mode
parameter was
not shown in the BridgeCreated event correctly.

Fixed it and added video_mode shown in the 'bridge show <bridge id>'
cli.

ASTERISK-28987

Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295