What's wrong with ast_debug?

  ast_debug is fine for general purpose debug output but it's not
  really geared for scope tracing since it doesn't present its
  output in a way that makes capturing and analyzing flow through
  Asterisk easy.

How is scope tracing better?

  Scope tracing uses the same "cleanup" attribute that RAII_VAR
  uses to print messages to a separate "trace" log level.  Even
  better, the messages are indented and unindented based on a
  thread-local call depth counter.  When output to a separate log
  file, the output is uncluttered and easy to follow.

  Here's an example of the output. The leading timestamps and
  thread ids are removed and the output cut off at 68 columns for
  commit message restrictions but you get the idea.

--> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
	--> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
		--> res_pjsip_session.c:3669 handle_incoming_response PJSIP/
			--> chan_pjsip.c:3265 chan_pjsip_incoming_response_after
				--> chan_pjsip.c:3194 chan_pjsip_incoming_response P
					    chan_pjsip.c:3245 chan_pjsip_incoming_respon
				<-- chan_pjsip.c:3194 chan_pjsip_incoming_response P
			<-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after
		<-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/
	<-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
<-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001

  The messages with the "-->" or "<--" were produced by including
  the following at the top of each function:

  SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session));

  Scope isn't limited to functions any more than RAII_VAR is.  You
  can also see entry and exit from "if", "for", "while", etc blocks.

  There is also an ast_trace() macro that doesn't track entry or
  exit but simply outputs a message to the trace log using the
  current indent level.  The deepest message in the sample
  (chan_pjsip.c:3245) was used to indicate which "case" in a
  "select" was executed.

How do you use it?

  More documentation is available in logger.h but here's an overview:

  * Configure with --enable-dev-mode.  Like debug, scope tracing
    is #ifdef'd out if devmode isn't enabled.

  * Add a SCOPE_TRACE() call to the top of your function.

  * Set a logger channel in logger.conf to output the "trace" level.

  * Use the CLI (or cli.conf) to set a trace level similar to setting
    debug level... CLI> core set trace 2 res_pjsip.so

Summary Of Changes:

  * Added LOG_TRACE logger level.  Actually it occupies the slot
    formerly occupied by the now defunct "event" level.

  * Added core asterisk option "trace" similar to debug.  Includes
	ability to specify global trace level in asterisk.conf and CLI
	commands to turn on/off and set levels.  Levels can be set
	globally (probably not a good idea), or by module/source file.

  * Updated sample asterisk.conf and logger.conf.  Tracing is
    disabled by default in both.

  * Added __ast_trace() to logger.c which keeps track of the indent
    level using TLS. It's #ifdef'd out if devmode isn't enabled.

  * Added ast_trace() and SCOPE_TRACE() macros to logger.h.
    These are all #ifdef'd out if devmode isn't enabled.

Why not use gcc's -finstrument-functions capability?

  gcc's facility doesn't allow access to local data and doesn't
  operate on non-function scopes.

Known Issues:

  The only know issue is that we currently don't know the line
  number where the scope exited.  It's reported as the same place
  the scope was entered.  There's probably a way to get around it
  but it might involve looking at the stack and doing an 'addr2line'
  to get the line number.  Kind of like ast_backtrace() does.
  Not sure if it's worth it.

Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027

Adds the "STIR_SHAKEN" dialplan function and an API call to add a
STIR_SHAKEN verification result to a channel. This information will be
held in a datastore on the channel that can later be queried through the
"STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results
including identity, attestation, and verify_result. Here are some
examples:

STIR_SHAKEN(count)
STIR_SHAKEN(0, identity)
STIR_SHAKEN(1, attestation)
STIR_SHAKEN(2, verify_result)

Getting the count can be used to iterate through the results and pull
information by specifying the index and the field you want to retrieve.

Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82

Add a new "masquarade" channel event, and use it in app_queue to track unique id's.

Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210

ASTERISK-28829 #close
ASTERISK-25844 #close

Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6

There are a lot of moving parts in this patch, but the focus of it is on
the verification of the signature using a public key located at the
public key URL provided in the JSON payload. First, we check the
database to see if we have already downloaded the key. If so, check to
see if it has expired. If it has, redownload from the URL. If we don't
have an entry in the database, just go ahead and download the public
key. The expiration is tested each time we download the file. After
that, read the public key from the file and use it to verify the
signature. All sanity checking is done when the payload is first
received, so the verification is complete once this point is reached.

The XML has also been added since a new config option was added to
general (curl_timeout). The maximum amount of time to wait for a
download can be configured through this option, with a low value by
default.

Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c

Some places in Asterisk did not treat the formats on a stream
as immutable when they are.

The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.

An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.

ASTERISK-28846

Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe

When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".

ASTERISK-28841

Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13

Allow voice volume to be multiplied or divided by a floating point number.

ASTERISK-28813

Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c

This unit test runs through combinations of...
	* Local codecs
	* Remote Codecs
	* Codec Preference
	* Incoming/Outgoing

A few new APIs were created to make it easier to test
the functionality but didn't result in any actual
functional change.

ASTERISK_28777

Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164

Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437

This change provides functions that take in a JSON payload, verify that
the contents contain all the mandatory fields and required values (if
any), and signs the payload with the private key. Four fields are added
to the payload: x5u, attest, iat, and origid. As of now, these are just
placeholder values that will be set to actual values once the logic is
implemented for what to do when an actual payload is received, but the
functions to add these values have all been implemented and are ready to
use. Upon successful signing and the addition of those four values, a
ast_stir_shaken_payload is returned, containing other useful information
such as the algorithm and signature.

Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393

named_acl.c (which is really a named_ha) now uses ast_ha_output.

I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set.  If this works then we can add
similar to other modules as required.

Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f

This fixes ast_addressfamily_to_sockaddrsize to reference the
provided argument, and ast_sockaddr_from_sockaddr to not use the name of
a structure as argument.

Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7

This commit sets up some of the initial framework for the module and
adds a way to read the private key from the specified file, which will
then be appended to the certificate object. This works fine for now, but
eventually some other structure will likely need to be used to store all
this information. Similarly, the caller_id_number is specified on the
certificate config object, but in the end we will want that information
to be tied to the certificate itself and read it from there.

A method has been added that will retrieve the private key associated
with the caller_id_number passed in. Tab completion for certificates and
stores has also been added.

Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f

Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5

Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f

When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.

Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.

ASTERISK-28755 #close

Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f

Update the state of remote_hold immediately on receipt of remote
SDP so that the information is available when building the SDP
answer

ASTERISK-28754 #close

Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f

There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.

Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e

There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.

Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113

This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.

This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.

ASTERISK-28697

Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b

This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.

The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.

Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.

ASTERISK-28733

Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5

When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.

A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.

ASTERISK-28716

Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867

In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use.  To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes"  In Asterisk versions <18, the default
is "no" preserving existing behavior.  Beginning with
Asterisk 18, the option will default to "yes".

NOTE:  This change does not affect UserEvents or the ARI
TextMessageReceived events.

* Added the "hide_messaging_ami_events" option to asterisk.conf.

* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
  the "Message/ast_msg_queue" channel if the option is set in
  asterisk.conf.  This suppresses the reporting of the events.

Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b

This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.

A description of the protocol can be found on the above referenced
GitHub page.  A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.

ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.

ASTERISK-28484 #close

Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5

ast_addressfamily_to_sockaddrize will determine the size that's
required, and ast_sockaddr_from_sockaddr then wraps this new function
and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without
knowing the address family) into the ast_sockaddr structure.

Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Some body generators, such as dialog-info+xml, require storing state
information which is then conveyed in the NOTIFY request itself. Up
until now there was no way for such body generators to persist this
information.

Two new API calls have been added to allow body generators to set and
get persisted data. This data is persisted out alongside the normal
persistence information and allows the body generator to restore
state information or to simply use this for normal storage of state.
State is stored in the form of JSON and it is up to the body
generator to interpret this as needed.

The dialog-info+xml body generator has been updated to take advantage
of this to persist the version number.

ASTERISK-27759

Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de

Adds source port matching support when IP matching is used:

  [example]
  type = identify
  match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444

If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.

ASTERISK-28639 #close
Reported by: Mitch Claborn

Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92

When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020

When a topic is added to a pool, its name has the pool's topic
name prepended.  For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020

The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.

When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.

Fix:

* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.

ASTERISK-28633
Reported by: Joeran Vinzens

Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6

When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.

ASTERISK-28562 #close
Reported by: Robert Sutton

Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d

This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.

ASTERISK-28629

Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc

Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.

ASTERISK-28604

Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7

ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee

A previous patch:

Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.

This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).

This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.

The default is disabled keeping with the old behavior.

ASTERISK-28660

Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a

Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies.  It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.

Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

* Pass caller information to frame allocation functions.
* Disable caching as it interfers with MALLOC_DEBUG reporting.
* Stop using ast_calloc_cache.

Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260

ASTERISK-28590 #close

Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c

Serializer pools have previously existed in Asterisk. However, for the most
part the code has been duplicated across modules. This patch abstracts the
code into an 'ast_serializer_pool' object. As well the code is now centralized
in serializer.c/h.

In addition serializer pools can now optionally be monitored by a shutdown
group. This will prevent the pool from being destroyed until all serializers
have completed.

Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971

Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they
both implemented their own serializer pool functionality that was pretty much
identical in each of the source files. This patch removes the duplicated code,
and uses the new 'ast_serializer_pool' object instead.

Additionally res_pjsip_mwi enables a shutdown group on the pool since if the
timing was right the module could be unloaded while taskprocessor threads still
needed to execute, thus causing a crash.

Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d

Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis

ASTERISK-28542 #close

Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8