Older versions of PJSIP do not have the proto field on the TLS transport
setting structure. This change adds a configure check so even if it is
not present we will still be able to build.

Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9

Not sure why it was there in the first place as we already specify
--disable-sound.

Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9

Configurations like "aors = a, b, c" were either ignoring everything after "a"
or trying to look up " b".  Same for mailboxes,  ciphers, contacts and a few
others.

To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip.  To
facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were
updated to handle null pointers.

In some cases, an ast_strlen_zero() test was added to skip consecutive commas.

There was also an attempt to ast_free an ast_strdupa'd string in
ast_sip_for_each_aor which was causing a SEGV.  I removed it.

Although this issue was reported for realtime, the issue was in the res_pjsip
modules so all config mechanisms were affected.

ASTERISK-25829 #close
Reported-by: Mateusz Kowalski

Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2

Change-Id: I265e4ac47c629c9a63dd86b59df82a7ab3c64384

Refactor and created function ast_cli_print_timestr_fromseconds to print
seconds formatted:  year(s) week(s) day(s) hour(s) second(s)

This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c,
res_config_ldap.c, res_config_pgsql.c.

Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c

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

Per RFC3325, the 'From' header is now anonymized on outgoing calls when
caller id presentation is prohibited.

TID = trust_id_outbound
PRO = Set(CALLERID(pres)=prohib)
USR = endpoint/from_user
DOM = endpoint/from_domain
PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)

Conditions          |Result
--------------------|----------------------------------------------------
TID PRO USR DOM     |PAI    FROM
--------------------|----------------------------------------------------
Y   Y   abc def.ghi |PRI    "Anonymous" <sip:abc@def.ghi>
Y   Y   abc         |PRI    "Anonymous" <sip:abc@anonymous.invalid>
Y   Y       def.ghi |PRI    "Anonymous" <sip:anonymous@def.ghi>
Y   Y               |PRI    "Anonymous" <sip:anonymous@anonymous.invalid>

Y   N   abc def.ghi |YES    <sip:abc@def.ghi>
Y   N   abc         |YES    <sip:abc@<ip_address>>
Y   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
Y   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

N   Y   abc def.ghi |NO     "Anonymous" <sip:abc@def.ghi>
N   Y   abc         |NO     "Anonymous" <sip:abc@anonymous.invalid>
N   Y       def.ghi |NO     "Anonymous" <sip:anonymous@def.ghi>
N   Y               |NO     "Anonymous" <sip:anonymous@anonymous.invalid>

N   N   abc def.ghi |YES    <sip:abc@def.ghi>
N   N   abc         |YES    <sip:abc@<ip_address>>
N   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
N   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

ASTERISK-25791 #close
Reported-by: Anthony Messina

Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9

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