Previously you could add [!dnid] to the SIP dial string to alter the To:
header. This change allows you to alter the From header as well.

SIP dial string extra options now look like this:

    [![touser[@todomain]][![fromuser][@fromdomain]]]

INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To:
header, that is no longer possible.

ASTERISK-25803 #close

Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7

When Asterisk receives a 412 (Conditional Request Failed) response
it has to recreate publish session.
There is bug in res_pjsip_outbound_publish.c
The function sip_outbound_publish_client_alloc is called with wrong object
while processing 412 (Conditional Request Failed) response.
This patch fixes it.

ASTERISK-25229 #close

Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359

The threadpool_auto_increment test fails infrequently for a couple of
reasons
* The threadpool listener was notified of fewer tasks being pushed than
  were actually pushed
* The "was_empty" flag was set to an unexpected value.

The problem is that the test pushes three tasks into the threadpool.
Test expects the threadpool to essentially gather those three tasks, and
then distribute those to the threadpool threads. It also expects that as
the tasks are pushed in, the threadpool listener is alerted immediately
that the tasks have been pushed. In reality, a task can be distributed
to the threadpool threads quicker than expected, meaning that the
threadpool has already emptied by the time each subsequent task is
pushed. In addition, the internal threadpool queue can be delayed so
that the threadpool listener is not alerted that a task has been pushed
even after the task has been executed.

From the test's point of view, there's no way to be able to predict
exactly the order that task execution/listener notifications will occur,
and there is no way to know which listener notifications will indicate
that the threadpool was previously empty.

For this reason, the test has been updated to only check the things it
can check. It ensures that all tasks get executed, that the threads go
idle after the tasks are executed, and that the listener is told the
proper number of tasks that were pushed.

Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c

The return type of ast_cel_track_event() is not large enough to return all
64 potential bits of the event enable mask.  Fortunately, the defined CEL
events do not really need all 64 bits and the return value is only used to
determine if the requested CEL event is enabled.

* Made the ast_cel_track_event() return 0 or 1 only so the return value
can fit inside an int type instead of zero or a truncated 64 bit non-zero
value.

Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c

Fix calculate of average time for talktime is wrong when is completed the
first call beacuse the time for talked would be that call.

ASTERISK-25800 #close

Change-Id: I94f79028935913cd9174b090b52bb300b91b9492

res_odbc.exports.in was missing a few symbols.
Changed to wildcards.

Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c

res_statsd.export.in was missing the _va variations of the log
functions causing Asterisk to crash in res_pjsip if OPTIONAL_API
wasn't enabled.

ASTERISK-25727 #close
Reported-by: Gergely Dömsödi

Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b

If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid
or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify
the header added by the dialplan function.  Since the header added by the
dialplan function is generic string, there are no virtual functions to parse
the uri and we get a segfault when we try.  Since the modify, was really only
an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER
and recreate it.

This raises a question for another time though:  What should happen with
duplicate headers?  Right now res_pjsip_header_funcs doesn't check for dups
so if it's session supplement is loaded after res_pjsip_caller_id's (or any
other module that adds headers), there'll be dups in the message.

ASTERISK-25337 #close

Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa

It is possible when processing a SIP REGISTER request to have two
threads end up creating contact_status structures in sorcery.
contact_status is created using a "find or create" function. If two
threads call into this at the same time, each thread will fail to find
an existing contact_status, and so both will end up creating a new
contact status.

During testing, we would see sporadic failures because the
PJSIP_CONTACT() dialplan function would operate on a different
contact_status than what had been updated by res_pjsip/pjsip_options.

The fix here is two-fold:
1) The "find or create" function for contact_status now has a lock
around the entire operation. This way, if two threads attempt the
operation simultaneously, the first to get there will create the object,
and the second will find the object created by the first thread.

2) res_sorcery_memory has had its create callback updated so that it
will not allow for objects with duplicate IDs to be created.

Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97

A problem arose when testing the AMI subscription listing actions where it
was possible for a subscription that had not been fully initialized to be
listed. This was problematic as the underlying listing code would crash.

This change makes it so the subscription tree is fully set up before it is
added to the list of subscriptions. This ensures that when the listing actions
get the subscription it is valid.

ASTERISK-25738 #close

Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48

load_module was just too hairy with every step having to clean up all
previous steps on failure.

Some of the pjproject init calls have now been moved to a separate
load_pjsip function and the unload_pjsip function was enhanced to clean
up everything if an error happened at any stage of the load process.

In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
and ast_threadpool_shutdowns were also corrected.

Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302

* heap-use-after-free happens when we free "cfg"
but then use "value" which refers to it

* A memory leak occurs because in some cases
it is not released "defaults"

ASTERISK-25721 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav

Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469

ASTERISK-25272 #close
Reported by: Etienne Lessard
patches:
 AST-25272.patch submitted by Etienne Lessard (license #6394)

Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17

Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
increments the lock on the returned dialog.  To account for this, configure.ac
now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
the original call or the new one.  If the new one was used, the ref count is
decremented before returning.

ASTERISK-25751 #close
Reported-by Josh Colp

Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8

Change-Id: I9290115a1aaadb589eb1d02eaeb502eec01b31fa

In older versions of the compiler was not sanitizes.
Compilers other than GCC can not support the Usan and TSAN
or have other options for *FLAGS.

ASTERISK-25767 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav

Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916

USAN can be used together with other sanitizers.

Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav

Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f

FD_SET contains a conditional statement to protect against buffer
overruns.  The statement was overly complicated and prevented use
of the last array element of ast_fdset.  We now just verify the fd
is less than ast_FDMAX.

Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40

When terminating the threads thrashing a sorcery memory cache each
would be told to stop and then we would wait on them. During at
least one thrashing test this was problematic due to the specific
usage pattern in use. It would take some time for termination of the
thread to occur.

This would occur due to contention between the threads retrieving
and the threads updating the cache. As the retrieving threads are
given priority it may be some time before the updating threads
are able to proceed.

This change makes it so all threads are told to stop and then each
are joined to ensure they stop. This way all the threads should
stop at around the same time instead of waiting for one to stop,
the next to stop, then the next, and so on. As a result of this
the execution time for each thrash test is much closer to their
expected value than previously seen as well.

Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827

Attempting to load a transport from realtime was forcing asterisk into an
infinite recursion loop.  The first thing transport_apply did was to do a
sorcery retrieve by id for an existing transport of the same name. For files,
this just returns the previous object from res_sorcery_config's internal
container, if any.  For realtime, the res_sourcery_realtime driver looks in the
database and finds the existing row but now it has to rehydrate it into a
sorcery object which means calling... transport_apply.  And so it goes.

The main issue with loading from realtime (apart from the loop) was that
transport stores structures and pointers directly in the ast_sip_transport
structure instead of the separate ast_transport_state structure.  This patch
separates those items into the ast_sip_transport_state structure.  The pattern
is roughly the same as res_pjsip_outbound_registration.

Although all current usages of ast_sip_transport and ast_sip_transport_state
were modified to use the new ast_sip_get_transport_state API, the original
items are left in ast_sip_transport and kept updated to maintain ABI
compatability for third-party modules.  They are marked as deprecated and
noted that they're now in ast_sip_transport_state.

ASTERISK-25606 #close
Reported-by: Martin Moučka

Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19

In case failed of command "realtime show pgsql status" show a message the data
of connection to more clear information in error.

Change-Id: Ia8e9e2400466606e7118f52a46e05df0719b6a29

Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98

A user cannot set new bridge options after the conference is created by
the first user.  Attempting to do so is documented as undefined behavior.

This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.

Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266

* changes:
  app_confbridge: Add ability to get the muted conference state.
  app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
  app_confbridge: Make non-admin users join a muted conference muted.

The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.

This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.

Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d

ps_systems needed disable_tcp_switch
ps_registrations needed line and endpoint

ASTERISK-25737 #close

Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19