RedHat/CentOS needs python-devel
Debian/Ubuntu needs automake, libsrtp-dev and python-dev

Ubuntu also needed libncurses5-dev for cmenuselect so while not
needed for pjproject, I adedd it anyway.

Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089

Apparently the != operator is fairly new so I've replaced it with
the old $(shell ...) syntax.

Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479
Reported-by: Richard Mudgett

Although we use the RTLD_LAZY flag when calling dlopen
the first time on a module, this only defers resolution
for function calls.  Pointer references to functions are
determined at link time so dlopen expects them to be there.
Since we don't cross-module link, pointers to functions
in other modules won't be available and dlopen will fail.

Doing a "hardened" build also causes problems because it
typically sets "-z now" on the ld command line which
overrides RTLD_LAZY at run time.

If the failing module isn't a GLOBAL_SYMBOLS module, then
dlopen will be called again after all the GLOBAL_SYMBOLS
modules have been loaded and they'll eventually resolve.

If the calling module IS a GLOBAL_SYMBOLS module itself
and a third module depends on it, then there's an issue
because the second time through the dlopen loop,
GLOBAL_SYMBOLS modules aren't given any special treatment
and since the order in which dlopen is called isn't
deterministic, the dependent may again be tried before the
module it needs is loaded.

Simple solution:  Save modules that fail load_resource
because of a dlopen error in a list and retry them
immediately after the first pass. Keep retrying until
the failed list is empty or we reach a #defined max
retries. Error messages are suppressed until the final
pass which also gets rid of those confusing error messages
about module failures that are later corrected.

Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb

It's possible for the transferer channel to get hung up early during the
attended transfer process. For instance, a phone may send a "bye" immediately
upon receiving a sip notify that contains a sip frag 100 (I'm looking at you
Jitsi). When this occurs a race begins between the transferer being hung up
and completion of the transfer code.

If the channel hangs up too early during a transfer involving stasis bridging
for instance, then when the created local channel goes to look up its swap
channel (and associated datastore) it can't find it (since it is no longer in
the bridge) thus it fails to enter the stasis application. Consequently, the
created local channel(s) hang up as well. If the timing is just right then the
bridging code attempts to add the message link with missing local channel(s).
Hence the crash.

Unfortunately, there is no great way to solve the problem of the unexpected
"bye". While we can't guarantee we won't receive an early hangup, and in this
case still fail to enter the stasis application, we can make it so asterisk
does not crash.

This patch does just that by locking the local channel structure, checking
that the local channel's peer has not been lost, and then continuing. This
keeps the local channel's peer from being ripped out from underneath it by
the local/unreal hangup code while attempting to set the stasis message link.

ASTERISK-25771

Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880

PJSIP does not ensure that when printing the message body the
buffer will be NULL terminated. This is problematic when searching
for the signal and duration values of the DTMF.

This change ensures the buffer is always NULL terminated.

Change-Id: I52653a1a60c93092d06af31a27408d569cc98968

Downgrade had a few issues.  First there was an errant 'update' statement in
add_auto_dtmf_mode that looks like it was a copy/paste error.  Second, we
weren't cleaning up the ENUMs so subsequent upgrades on postgres failed
because the types already existed.

For sqlite...  sqlite doesn't support ALTER or DROP COLUMN directly.
Fortunately alembic batch_operations takes care of this for us if we
use it so the alter and drops were converted to use batch operations.

Here's an example downgrade:

    with op.batch_alter_table('ps_endpoints') as batch_op:
        batch_op.drop_column('tos_audio')
        batch_op.drop_column('tos_video')
        batch_op.add_column(sa.Column('tos_audio', yesno_values))
        batch_op.add_column(sa.Column('tos_video', yesno_values))
        batch_op.drop_column('cos_audio')
        batch_op.drop_column('cos_video')
        batch_op.add_column(sa.Column('cos_audio', yesno_values))
        batch_op.add_column(sa.Column('cos_video', yesno_values))

    with op.batch_alter_table('ps_transports') as batch_op:
        batch_op.drop_column('tos')
        batch_op.add_column(sa.Column('tos', yesno_values))
    # Can't cast integers to YESNO_VALUES, so dropping and adding is required
        batch_op.drop_column('cos')
        batch_op.add_column(sa.Column('cos', yesno_values))

Upgrades from base to head and downgrades from head to base were tested
repeatedly for postgresql, mysql/mariadb, and sqlite3.

Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8

ast_sip_get_transport_states was returning a container of internal_state
objects instead of ast_sip_transport_state objects.  This was causing
transport lookups to fail, most noticably in res_pjsip_nat, which
couldn't find the correct external addresses.  This was causing contacts
to go out with internal ip addresses.

ASTERISK-25830 #close
Reported-by: Sean Bright

Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e

In message.c, if msg_alloc fails to init the string field,
vars may be null, so use a null tolerant cleanup.

In res_pjsip_messaging.c, if msg_data_create fails, mdata
will be null, so use a null tolerant cleanup.

ASTERISK-25323

Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56

This patch avoids crashing on a null pointer
if the strdup() allocation fails.

ASTERISK-25323

Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5

Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386

Previous chan_sip behavior:

Before this patch chan_sip would always strip any quotes from an incoming
reason and pass that value up as the REDIRECTING(reason).  For an outgoing
reason value, chan_sip would check the value against known values and
quote any it didn't recognize.  Incoming 480 response message reason text
was just assigned to the REDIRECTING(reason).

Previous chan_pjsip behavior:

Before this patch chan_pjsip would always pass the incoming reason value
up as the REDIRECTING(reason).  For an outgoing reason value, chan_pjsip
would send the reason value as passed down.

With this patch:

Both channel drivers match incoming reason values with values documented
by REDIRECTING(reason) and values documented by RFC5806 regardless of
whether they are quoted or not.  RFC5806 values are mapped to the
equivalent REDIRECTING(reason) documented value and is set in
REDIRECTING(reason).  e.g., an incoming RFC5806 'unconditional' value or a
quoted string version ('"unconditional"') is converted to
REDIRECTING(reason)'s 'cfu' value.  The user's dialplan only needs to deal
with 'cfu' instead of any of the aliases.

The incoming 480 response reason text supported by chan_sip checks for
known reason values and if not matched then puts quotes around the reason
string and assigns that to REDIRECTING(reason).

Both channel drivers send outgoing known REDIRECTING(reason) values as the
unquoted RFC5806 equivalent.  User custom values are either sent as is or
with added quotes if SIP doesn't allow a character within the value as
part of a RFC3261 Section 25.1 token.  Note that there are still
limitations on what characters can be put in a custom user value.  e.g.,
embedding quotes in the middle of the reason string is silly and just
going to cause you grief.

* Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
'cfu' value.

* Added missing malloc() NULL return check in res_pjsip_diversion.c
set_redirecting_reason().

* Fixed potential read from a stale pointer in res_pjsip_diversion.c
add_diversion_header().  The reason string needed to be copied into the
tdata memory pool to ensure that the string would always be available.
Otherwise, if the reason string returned by reason_code_to_str() was a
user's reason string then the string could be freed later by another
thread.

Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87

Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd

* Fix double unref of other_party channel in off nominal path.

* This is unlikely to be a real problem.  However, for safety,
in handle_incoming_request() keep the datastore ref with the
other_party channel ref until we are finished with the other_party
channel.

Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821

Background here:
http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html

From CHANGES:
 * To help insure that Asterisk is compiled and run with the same known
   version of pjproject, a new option (--with-pjproject-bundled) has been
   added to ./configure.  When specified, the version of pjproject specified
   in third-party/versions.mak will be downloaded and configured.  When you
   make Asterisk, the build process will also automatically build pjproject
   and Asterisk will be statically linked to it.  Once a particular version
   of pjproject is configured and built, it won't be configured or built
   again unless you run a 'make distclean'.

   To facilitate testing, when 'make install' is run, the pjsua and pjsystest
   utilities and the pjproject python bindings will be installed in
   ASTDATADIR/third-party/pjproject.

   The default behavior remains building with the shared pjproject
   installation, if any.

Building:

   All you have to do is include the --with-pjproject-bundled option on
   the ./configure command line (and remove any existing --with-pjproject
   option if specified).  Everything else is automatic.

Behind the scenes:

   The top-level Makefile was modified to include 'third-party' in the
   list of MOD_SUBDIRS.

   The third-party directory was created to contain any third party
   packages that may be needed in the future.  Its Makefile automatically
   iterates over any subdirectories passing on targets.

   The third-party/pjproject directory was created to house the pjproject
   source distribution.  Its Makefile contains targets to download, patch
   configure, generate dependencies, compile libs, apps and python bindings,
   sanitized build.mak and generate a symbols list.

   When bootstrap.sh is run, it automatically includes the configure.m4
   file in third-party/pjproject.  This file has a macro to download and
   conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR
   and PJPROJECT_BUNDLED.  It also tests for the capabilities like
   PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to
   trying to compile.  Of course, bootstrap.sh is only run once and the
   configure file is incldued in the patch.

   When configure is run with the new options, the macro in configure.m4
   triggers the download, patch, conifgure and tests.  No compilation is
   performed at this time.  The downloaded tarball is cached in /tmp so
   it doesn't get downloaded again on a distclean.

   When make is run in the top-level Asterisk source directory, it will
   automatically descend all the subdirectories in third_party just as it
   does for addons, apps, etc.  The top-level Makefile makes sure that
   the 'third-party' is built before 'main' so that dependencies from the
   other directories are built first.

   When main does build, a new shared library (libasteriskpj) is created that
   links statically to the pjproject .a files and exports all their symbols.
   The asterisk binary links to that, just as it does with libasteriskssl.

   When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject
   python bindings are installed in ASTDATADIR/third-party/pjproject.  This
   will facilitate testing, including running the testsuite which will be
   updated to check that directory for the pjsua module ahead of the system
   python library.

Modules should continue to depend on pjproject if they use pjproject APIs
directly.  They should not care about the implementation.  No changes to any
res_pjsip modules were made.

Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103

chan_sip could not handle AST_T38_TERMINATED frames being sent to it when
the channel left the bridge.  The action resulted in overlapping outgoing
reINVITEs.  The testsuite tests/fax/sip/directmedia_reinvite_t38 was not
happy.

* Force T.38 to be remembered as locally bridged.  Now when the channel
leaves the native RTP bridge after T.38, the channel remembers that it has
already reINVITEed the media back to Asterisk.  It just needs to terminate
T.38 when the AST_T38_TERMINATED arrives.

* Prevent redundant AST_T38_TERMINATED from causing problems.  Redundant
AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if
they happen before the T.38 state changes to disabled.  Now the T.38 state
is set to disabled before the reINVITE is sent.

ASTERISK-25582 #close

Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce

This backs out item 4 of the 4875e5ac
commit.  Item 4 added the t38_bye_supplement.  Unfortunately, the frame
that it puts into the bridge may or may not be processed by the time the
bridged peer is kicked out of the bridge.  If it is processed then all is
well.  However, if it is not processed then that channel is stuck in fax
mode until it hangs up or maybe if it joins another bridge for T.38
faxing.

ASTERISK-25582

Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7

The channel is now going to get T.38 terminated when it leaves the
bridging system and the bridged peers are going to get T.38 terminated as
well.

ASTERISK-25582

Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7

ASTERISK-25582

Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b

Local channel optimization could cause DTMF digits to be duplicated.
Pending DTMF end events would be posted to a bridge when the local channel
optimizes out and is replaced by the channel further down the chain.  When
the real digit ends, the channel would get another DTMF end posted to the
bridge.

A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B

1) LocalA has the /n flag to prevent optimization.
2) B is sending DTMF to A through the local channel chain.
3) When LocalB optimizes out it can move B to the position of LocalB;1
4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
settle an owed DTMF end to the bridge toward LocalA;2.
5) When B finally ends its DTMF it sends the DTMF end down the chain.
6) Without this patch, A would hear the DTMF digit end when LocalB
optimizes out and when B ends the original digit.

ASTERISK-25582

Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251

Frame hooks can conceivably return a control frame in exchange for an
audio frame inside ast_write().  Those returned control frames were not
handled quite the same as if they were sent to ast_indicate().  Now it
doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a
channel or ast_indicate().

ASTERISK-25582

Change-Id: I5775f41421aca2b510128198e9b827bf9169629b

The ast_sorcery_create, update and delete function have been refactored
to better deal with caches and errors.

The action is now called on all non-caching wizards first. If ANY succeed,
the action is called on all caching wizards and the observers are notified.
This way we don't put something in the cache (or update or delete) before
knowing the action was performed in at least 1 backend and we only call the
observers once even if there were multiple writable backends.

ast_sorcery_create was never adding to caches in the first place which
was preventing contacts from getting added to a memory_cache when they
were created.  In turn this was causing memory_cache to emit errors if
the contact was deleted before being retrieved (which would have
populated the cache).

ASTERISK-25811 #close
Reported-by: Ross Beer

Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46