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

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

ast_trace() was always emitting messages when it's level was set to -1
because it was ignoring scope_level.

Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506

Implemention of History-Info capable of interworking with Diversion
Header following RFC7544

ASTERISK-29027 #close

Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6

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

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

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

The assumed behavior of realloc() - that it was effectively a free() if
its second argument was 0 - is Linux specific behavior and is not
guaranteed by either POSIX or the C specification.

Instead, if we want to resize a vector to 0, do it explicitly.

Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f

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

This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.

Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d

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

Allows a vector to be sorted in-place, rather than only during
insertion.

Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780

* Updated AMI version to 8.0.0
* Updated ARI version to 7.0.0
* Update make_ari_stubs.py to "Asterisk 19"

Change-Id: I51fb38c2e29f2db785f64a8bbd5565d56bea5af5

Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.

Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.

Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.

Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1

Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c

This commit adds the endpoint options required to control
Advanced Codec Negotiation.

incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs

The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs.  That'll be handled
when things finalize.

This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.

Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11

 * ast_frame_subclass2str() and ast_frame_type2str() now return
   a pointer to the buffer that was passed in instead of void.
   This makes it easier to use these functions inline in
   printf-style debugging statements.

 * Added many missing control frame entries in
   ast_frame_subclass2str.

Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc

Tracing through synchronous tasks was a little troublesome because
the new thread's stack counter reset to 0.  This change allows
a synchronous task to set its trace level to be the same as the
thread that pushed the task.  For now, the task's level has to be
passed in the task's data structure but a future enhancement to the
taskprocessor subsystem could automatically set the trace level
of the servant to be that of the caller.

This doesn't really make sense for async tasks because you never
know when they're going to run anyway.

Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825

This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.

Note, the AMI version has been bumped for this change.

ASTERISK-28945 #close

Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb