__ast_mutex_logger used the variable `canlog` without accepting it as a
argument.  Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement.  This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.

Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853

Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push.  This will help
ensure we perform the necessary cleanup upon failure.

Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d

This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts
startup, but it's possible for this code to be returned on manual load
of a module after startup.

It is an error for a module to not have a load callback but this is not
a fatal system error.  In this case flag the module as declined, return
AST_MODULE_LOAD_FAILURE only if a required module is broken.

Expand doxygen documentation for AST_MODULE_LOAD_*.

Change-Id: I3c030bb917f6e5a0dfd9d91491a4661b348cabf8

* Display list of unavailable dependencies when they cause another
  module to fail loading.
* When a module declines to load find all modules which depend on it so
  they can be declined and listed together.
* Prevent retry of declined modules during startup.
* When a module fails to dlopen try loading it with RTLD_LAZY so we can
  attempt to display the list of missing dependencies.

These changes are meant to reduce logger spam that is caused when a
module has many dependencies and declines to load.  This also fixes some
error paths which failed to recognize required modules.

Module load/start errors are delayed until the end of loader startup.

Change-Id: I046052c71331c556c09d39f47a3b92975f3e1758

Add a volatile flag to lock tracking structures so we only need to use
the global lock when first initializing tracking.

Additionally add support for DEBUG_THREADS_LOOSE_ABI.  This is used by
astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is
not defined.

Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b

In order to do this and provide good feedback, a new macro was
created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and
path setups for the library then compiles, links and runs a supplied
code fragment to do the final determination.  In this case, the
final code fragment compares UNBOUND_VERSION_MAJOR
and UNBOUND_VERSION_MINOR to determine if they're greater than or
equal to 1.5.

Since we require version 1.5, some code in res_resolver_unbound
was also simplified.

ASTERISK-28045
Reported by: Samuel Galarneau

Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72

Use json_vsprintf from versions which contain fix for va_copy leak.

Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.

Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539

When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.

This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.

Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.

Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0

There's been a long standing leak when using topic pools.  The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically.  If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.

* Added stasis_topic_pool_delete_topic() so modules can clean up
  topics from pools.
* Registered the topic pool containers so it can be examined from
  the CLI when AO2_DEBUG is enabled.  They'll be named
  "<topic_pool_name>-pool".

Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25

This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.

ASTERISK-28059

Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189

Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df

Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.

* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
  pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
  output.

* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
  in brackets.

* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
  to also set pjsip_rx_data.pkt_info.src_addr.

Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8

Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address
needs to be rendered to a string and then parsed into the correct
structure. This also involves a call to getaddrinfo(3). The same is true
for the inverse operation.

Instead, because we know the internal structure of both ast_sockaddr and
pj_sockaddr, we can translate directly between the two without the
need for an intermediate string.

Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761

Change-Id: I9cf97bdc756012d1f552ab007f4aa85e0ddb4e62

When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.

Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37

* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)

Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7

Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree.  Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h".  In most cases each option
only applies to one or two files.  Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds.  For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.

Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename.  Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.

The only code change is a slight adjustment to how main/astmm.c is
initialized.  Initialization functions always exist so main/asterisk.c
can call them unconditionally.  Additionally rename the astmm
initialization functions so they are not exported.

Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027

In Solaris, the header <jansson.h> is in /usr/include/jansson. To find
Jansson even in such a subdirectory, the tool pkg-config is queried via
AST_PKG_CONFIG_CHECK. For those platforms, which do not list Jansson via
pkg-config, the previous check remains and is executed thereafter.

Because the check for the NetBSD Editline library uses the tool pkg-config
the code of PKG_PROG_PKG_CONFIG must be used. Because that check happens
earlier than Jansson, it must be placed in front of that.

ASTERISK-27991

Change-Id: I69ea0f379f87a50049654b2487c76ee1c04fa53a

When publishing a device state the change can be marked as being
cachable or not. If it is not cached the change is just published
to all interested and not stored away for later query. This was not
fully taken into account when publishing in stasis. The act of
publishing would create a topic for the device even if it may be
ephemeral.

This change makes it so messages which are not cached won't create
a topic for the device. If a topic does already exist it will be
published to but otherwise the change will only be published to
the device state all topic.

ASTERISK-27591

Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a

The "xmldoc dump" cli command was simply concatenating xml documents
into the output file.  The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.

* Added 2 new functions to xml.c:
    ast_xml_copy_node_list creates a copy of a list of children.
    ast_xml_add_child_list adds a list to an existing list.

* Modified handle_dump_docs to create a new output document and
  add to it the children from each input file.  It then dumps the
  new document to the output file.

Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07

In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.

This assertion is no longer present so the special logic can be
removed to make it the same between them both.

Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8

Change-Id: Ib3111b151d37cbda40768cf2a8a9c6cf6c5c7cbd

Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.

If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.

According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.

Also added additional functionality to ast_data_buffer, along with some
testing.

For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements

ASTERISK-27810 #close

Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d

* Merge the preload and load stages, use load ordering to try preload's
  first.  This fixes an issue where `preload=res_config_curl` would fail
  unless res_curl and func_curl were also preloaded.  Now it is only
  required that those modules be loaded during startup: autoload or
  regular load is good enough.
* The configuration option `require` and `preload-require` were only
  effective if the modules failed to load.  These options will now abort
  Asterisk startup if required modules fail to reach the 'Running'
  state.
* Missing or invalid 'module.conf' did not prevent startup.  Asterisk
  doesn't do anything without modules so this a fatal error.

Change-Id: Ie4176699133f0e3a823b43f90c3348677e43a5f3

Keep track if ICE candidates were in the SDP offer & only put them
in the corresponding SDP answer if the offer condaind ICE candidates

ASTERISK-27957 #close

Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92

A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.

ASTERISK-27949
Reported-by: Ross Beer

Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1

The AMI action was directly sending the text to the channel driver.
However, this makes two threads attempt to handle media and runs afowl of
CHECK_BLOCKING.

* Queue a read action to make the channel's media handling thread actually
send the text message.  This changes the AMI actions success/fail response
to just mean the text was queued to be sent not that the text actually got
sent.  The channel driver may not even support sending text messages.

ASTERISK-27943

Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379

pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response.  We handle this correctly.  There have
been reported cases where the To tag is the same but we still need to
follow the media.  The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime.  The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.

So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.

The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.

Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer

ASTERISK-27933
Reported by: bautsche

Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49

* Removed an unnecessary call to ast_channel_blocker_set() in
__ast_read().

ASTERISK-27625

Change-Id: I342168b999984666fb869cd519fe779583a73834

There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.

ASTERISK-27625

Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e

There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.

ASTERISK-27625

Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905

The CHECK_BLOCKING() macro is used to indicate if a channel's handling
thread is about to do a blocking operation (poll, read, or write) of
media.  A few operations such as ast_queue_frame(), soft hangup, and
masquerades use the indication to wake up the blocked thread to reevaluate
what is going on.

ASTERISK-27625

Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d

Executing dialplan functions from either AMI or ARI by getting a variable
could place the channel into autoservice.  However, these user interface
threads do not handle the channel's media so we wind up with two threads
attempting to handle the media.

There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.

ASTERISK-27625

Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49

Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8

It is invalid to typedef something more than once.  Though not all gcc
compilers on different OS's complain about it.

Change-Id: I5a7d4565990c985822d61ce75bde0b45f9870540

Furthermore, allow OpenSSL configured with no-dh. Additionally, this change
allows auto-negotiation of the elliptic curve/group for servers, not only with
OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519
(since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect.

ASTERISK-27910

Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537

asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those
inclusions got replaced by forward declarations. As side effect, the inclusions
got completed.

ASTERISK-27878

Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7

When RTP was originally created it had the ability to place a single
extension in an RTP packet. In practice people wanted to potentially
put multiple extensions in one and so RFC 5285 (obsoleted by RFC
8285) came into existence. This allows RTP extensions to be negotiated
with a unique identifier to be used in the RTP packet, allowing
multiple extensions to be present in the packet.

This change extends the RTP engine API to add support for this. A
user of it can enable extensions and the API provides the ability to
retrieve the information (to construct SDP for example) and to provide
negotiated information (from SDP). The end result is that the RTP
engine can then query to see if the extension has been negotiated and
what unique identifier is to be used. It is then up to the RTP engine
implementation to construct the packet appropriately.

The first extension to use this support is abs-send-time which is
defined in the REMB draft[1] and is a second timestamp placed in an
RTP packet which is for when the packet has left the sending system.
It is used to more accurately determine the available bandwidth.

ASTERISK-27831

[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03

Change-Id: I508deac557867b1e27fc7339be890c8018171588

This function originally was used in chan_sip to enable some simplifying
assumptions and eventually was copy and pasted into res_pjsip_logger and
res_hep.  Since it's replicated in three places, it's probably best to
move it into the public netsock2 API for these modules to use.

Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04