gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.

ASTERISK-26157 #close

Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e

This patch removes the following modules:
 - pbx_functions: It never existed.
 - res_pjsip_log_forwarder: It no longer exists.
 - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
                  aren't going to be installing HOMER
 - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
                  loaded, and we aren't configured to make use of the
                  module

Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5

fax_v21_session_new created a session details object but only released
the allocation reference during error conditions.  fax_session_new adds
it's own reference to details if needed so the caller is always
responsible for cleaning it's own reference.

ASTERISK-26141 #close

Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88

gcc 6 caught a previously unidentified self-comparison in
ice_candidate_cmp.  Fixed it and re-ordered the predicates for better
short-circuiting.

ASTERISK-26140 #close

Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7

A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
was using a pointer to a pointer as the destination of a memcpy and a
'&' instead of '*' in the sizeof.

ASTERISK-26138 #close

Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708

A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.

In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.

This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.

ASTERISK-26128 #close

Change-Id: I51574736a881189de695a824883a18d66a52dcef

Since the file was missing the depends on pjproject, it wasn't
picking up the pjproject related include path.  If there was no
system installed pjproject and pjproject-bundled was used, a compile
would fail because pjsip.h wasn't found.

ASTERISK-26139 #close

Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757

Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not
support the platform SVR2 from the year 1987 anymore.

ASTERISK-26046

Change-Id: I28161b037feb2d29ab46ed20e785928460226c22

ensure that cert bios get freed after creating the fingerprint

ASTERISK-26129 #close

Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451

Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c

Occasionally under load we'll attempt to send a final NOTIFY on a
subscription that's already been terminated and a SEGV will occur
down in pjproject's evsub_destroy function.  This is a result of a
race condition between all the paths that can generate a notify
and/or destroy the underlying pjproject evsub object:

 * The client can send a SUBSCRIBE with Expires: 0.
 * The client can send a SUBSCRIBE/refresh.
 * The subscription timer can expire.
 * An extension state can change.
 * An MWI event can be generated.
 * The pjproject transaction timer (timer_b) can expire.

Normally when our pubsub_on_evsub_state is called with a terminate,
we push a task to the serializer and return at which point the dialog
is unlocked.  This is usually not a problem because the task runs
immediately and locks the dialog again.  When the system is heavily
loaded though, there may be a delay between the unlock and relock
during which another event may occur such as the subscription timer
or timer_b expiring, an extension state change, etc.  These may also
cause a terminate to be processed and if so, we could cause pjproject
to try to destroy the evsub structure twice.  There's no way for us to
tell that the evsub was already destroyed and the evsub's group lock
can't tolerate this and SEGVs.

The remedy is twofold.

 * A patch has been submitted to Teluu and added to the bundled
   pjproject which adds add/decrement operations on evsub's group lock.

 * In res_pjsip_pubsub:
   * configure.ac and pjproject-bundled's configure.m4 were updated
     to check for the new evsub group lock APIs.
   * We now add a reference to the evsub group lock when we create
     the subscription and remove the reference when we clean up the
     subscription.  This prevents evsub from being destroyed before
     we're done with it.
   * A state has been added to the subscription tree structure so
     termination progress can be tracked through the asyncronous tasks.
   * The pubsub_on_evsub_state callback has been split so it's not doing
     double duty.  It now only handles the final cleanup of the
     subscription tree.  pubsub_on_rx_refresh now handles both client
     refreshes and client terminates.  It was always being called for
     both anyway.
   * The serialized_on_server_timeout task was removed since
     serialized_pubsub_on_rx_refresh was almost identical.
   * Missing state checks and ao2_cleanups were added.
   * Some debug levels were adjusted to make seeing only off-nominal
     things at level 1 and nominal or progress things at level 2+.

ASTERISK-26099 #close
Reported-by: Ross Beer.

Change-Id: I779d11802cf672a51392e62a74a1216596075ba1

Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the
underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for
WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based
cipher-suites.

ASTERISK-26130 #close

Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0

The receipt of a SIP MESSAGE may occur over any transport including TCP
and TLS. When the message is received, the original URI is added to the
message in the field PJSIP_RECVADDR, but this is insufficient to ensure
a reply message can reach the originating endpoint. This patch adds the
PJSIP_TRANSPORT field populated with the transport type.

ASTERISK-26132 #close

Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e

Some configure scripts used both AC_HELP_STRING and its replacement
AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were
changed to AS_HELP_STRING.

ASTERISK-26046

Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f

When shutting down res_pjsip_session will get unloaded before res_pjsip.
The act of unloading unregisters all the PJSIP services and sets
their module IDs to -1. In some cases it is possible for a timer to
occur after this happens which calls into res_pjsip_session. The
res_pjsip_session module can then try to get the session from the
INVITE session using the module ID. Since the module ID is now -1
this fails.

This change stores a copy of the module ID and uses it for the timer
callback scenario. If the module ID is -1 the callback immediately
returns but if the module ID is valid then it continues as normal.

This works as the original ID of the module is guaranteed to still
be valid when used with the INVITE session.

ASTERISK-26127 #close

Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573

Fix compile error introduced by the patch for
ASTERISK-26045

Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3

ASTERISK-26119 #close

Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c

Announcer channels were not being destroyed because the
stasis_app_control structure that referenced them was not being
destroyed. The control structure was not being destroyed because it was
not being unlinked from its container. It was not being unlinked from
its container because the after bridge callback for the announcer
channel was not being run. The after bridge callback was not being run
because the after bridge datastore was not being removed from the
channel on destruction. The channel was not being destroyed because the
hangup that used to destroy the channel was now only reducing the
reference count to one. The reference count of the channel was only
being reduced to one because the stasis_app_control structure was
holding the final reference...

The control structure used to not keep a reference to the channel, so
that loop described above did not happen.

The solution is to manually remove the control structure from its
container when the playback on a bridge is complete.

ASTERISK-26083 #close
Reported by Joshua Colp

Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4

The internal HTTP/WebSocket server supports both TCP and TLS, which can be
activated separately via the file http.conf. The source code intends to re-use
the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified
explicitly. This did not work because of a typo. This change resolves this typo.

ASTERISK-26126 #close

Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f

* In unload_module(), reordered destroying things to minimize the window
that the global transports container could be used by other threads on
shutdown.  When shutting down you need to stop things in the opposite
order of creation.

* Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to
eliminate the crash potential by other threads using the container on
shutdown.

* Made struct monitored_transport.sip_received not use
ast_atomic_fetchadd_int() since it is used as a boolean value that is only
set TRUE.  It was previously incremented for every received SIP message
and could theoretically overflow.

* In monitored_transport_state_callback(), allocated the monitored
transport object without a lock since the lock was unused.

* In keepalive_global_loaded(), removed releasing the transports container
if the keepalive_thread could not be started.  I set it up to be tried
again if the user reloads the configuration.

Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff

Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1

Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3