Review: https://reviewboard.asterisk.org/r/2805/
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The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
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(issue ASTERISK-22145)
(closes issue ASTERISK-22145)
Reported By: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2811/
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Some configuration objects currently won't place nice if reloaded.
Specifically, in this case the pjsip transport objects.  Now when
registering an object in sorcery one may specify that the object is
allowed to be reloaded or not.  If the object is set to not reload
then upon reloading of the configuration the objects of that type
will not be reloaded.  The initially loaded objects of that type
however will remain.

While the transport objects will not longer be reloaded it is still
possible for a user to configure an endpoint to an invalid transport.
A couple of log messages were added to help diagnose this problem if
it occurs.

(closes issue ASTERISK-22382)
Reported by: Rusty Newton
(closes issue ASTERISK-22384)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2807/
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main/config.c - cleanup cache fie includes
res/res_security_log.c - unregister logger level
channesl/chan_sip.c - cleanup io context and notify_types
main/translator.c - cleanup at shutdown
main/named_acl.c - cleanup cli commands
main/indications.c - ast_get_indication_tone() unref default_tone_zone if used

(closes issues ASTERISK-22378)
Reported by: Corey Farrell
Patches:
     config_shutdown.patch uploaded by coreyfarrell (license 5909)
     res_security_log.patch uploaded by coreyfarrell (license 5909)
     chan_sip-11.patch uploaded by coreyfarrell (license 5909)
     indications_refleak.patch uploaded by coreyfarrell (license 5909)
     named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license 5909)
     translate_shutdown.patch uploaded by coreyfarrell (license 5909)

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This simply pulls in the changes that were breaking from the CHANGES file
and updates a few other areas accordingly. It also removes the 10 => 11
notes, which are traditionally removed from each major version and stored
in the appropriate UPGRADE-X.txt file.
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Parkinglots are defined in res_features.conf now, but this patch fixes
features_config so that features don't fail to load when parkinglots
are present in features.conf

Review: https://reviewboard.asterisk.org/r/2801/
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(closes issue ASTERISK-22426)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2806/
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ast_xmldoc_printable returns an allocated block that must be freed by the
caller.  Fixed manager.c and res_agi.c to stop leaking these results.

(closes issue ASTERISK-22395)
Reported by: Corey Farrell
Patches:
     manager-leaks-12.patch uploaded by coreyfarrell (license 5909)
     res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909)
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Revert the -r392190 change.  The original test was correct.  The CDR code
was actually returning an unititialized buffer.
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You cannot put the "Testing <blah> pass/fail" on a single line before
actually performing the test.  Now any additional failure information is
logged before the test pass/fail announcement.

* Added an additional CDR(answer,u) test.
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(closes issue ASTERISK-22368)
Reported by: Corey Farrell
Patches:
     issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes (license 5674)
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With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].

This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.

For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.

Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)

The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.

Other changes made as a part of this patch:
 * The stubs for http_websocket that wrap system calls set errno to
   ENOSYS.

 * res_http_websocket now properly increments module use count.

 * In loader.c, the while() wrappers around dlclose() were removed. The
   while(!dlclose()) is actually an anti-pattern, which can lead to
   infinite loops if the module you're attempting to unload exports a
   symbol that was directly linked to.

 * The special handling of nonoptreq on systems without weak symbol
   support was removed, since we no longer rely on weak symbols for
   optional_api.

 [1]: https://wiki.asterisk.org/wiki/x/wACUAQ

(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.

The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.

There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.

The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)

(closes issue ASTERISK-21582)
Review: https://reviewboard.asterisk.org/r/2693/
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  r397975 | rmudgett | 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line
  
  pbx.c: Make ast_str_substitute_variables_full() not mask variables.
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  r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013) | 1 line
  
  Revert last commit.
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PJSIP's PIDF API does not replace angle brackets with
their appropriate counterparts for XML. So we have to
do it ourself. In this particular case, the problem had
to do with attempting to place an unsanitized SIP URI
into an XML node. Now we don't get a 488 from recipients
of our PIDF NOTIFYs.
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The previous method did not allocate enough space to create
the entire string, but adjusted the string's slen value to
be larger than the actual allocation. This resulted in garbled
text in NOTIFY requests from Asterisk.

This method allocates the proper amount of space first and then
writes the content into the buffer.
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Refactored cases where a combination of ast_verbose/options_verbose were
present.  Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used.  Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.

(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
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The previous placement would result in the resubscribe() callback called instead of
the subscription_terminated() callback being called when a subscription was ended
via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw
an assertion.
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RFC 5407 section 3.1.2 details a scenario where a UAC sends
a CANCEL at the same time that a UAS sends a 200 OK for the
INVITE that the UAC is canceling. When this occurs, it is the
role of the UAC to immediately send a BYE to terminate
the call.

This scenario was reproducible by have a Digium phone with two lines
place a call to a second phone that forwarded the call to the second
line on the original phone. The Digium phone, upon realizing that it
was connecting to itself, would attempt to cancel the call. The timing
of this happened to trigger the aforementioned race condition about
80% of the time. Asterisk was not doing its job of sending a BYE
when receiving a 200 OK on a cancelled INVITE. The result was that
the ast_channel structure was destroyed but the underlying SIP
session, as well as the PJSIP inv_session and dialog, were still
alive. Attempting to perform an action such as a transfer, once in
this state, would result in Asterisk crashing.

The circumstances are now detected properly and the session is ended
as recommended in RFC 5407.

(closes issue AST-1209)
reported by John Bigelow
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(closes ASTERISK-22376)
Reported by: John Hardin
Patches:
     memleak.patch uploaded by jhardin (license 6512)
     memleak2.patch uploaded by jhardin (license 6512)
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The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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Without this, documentation defined in sub-folders is ignored. Since having
properly generated documentation is especially important in Asterisk 12 -
not having it can cause a module to not load - 'make full' needs to look in
all .c files.
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  r397921 | mmichelson | 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines
  
  Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events.
  
  Attempting to transfer an unbridged call would result in crashes in either CEL code or
  in the conversion to AMI messages.
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  r397922 | mmichelson | 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) | 3 lines
  
  Remove extra debug message.
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In r397874, the scripts were removed... but not replaced. Thanks to
Michael Young for noticing this!
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* Made ast_strftime_locale() ensure that the output buffer is initialized.
The std library strftime() returns 0 and does not touch the buffer if it
has an error.  However, the function can also return 0 without an error.

(closes issue ASTERISK-22412)
Reported by: rmudgett
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* Fixed return value of ast_cdr_serialize_variables() on error.  It needs
to return 0 indicating no CDR variables found.

* Made ast_cdr_serialize_variables() check the return value of
cdr_object_format_property() and assert if nonzero.  A member of the
cdr_readonly_vars[] was not handled.

* Removed unused elements from cdr_readonly_vars[]: total_duration,
total_billsec, first_start, and first_answer.
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A problem encountered during testing was that res_pjsip_refer would
not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook
that was supposed to send the NOTIFY would never be told that an answer
had occurred. This happened for two reasons:

1) The transferee channel on which the framehook was on was already up.
2) Answers are rarely if ever written to channels. Rather, the ast_answer()
or ast_raw_answer() function is used to answer channels.

Thanks to a suggestion by Matt Jordan, the best way to detect that the call
had been answered was to find out when the transferee channel joined a bridge.
With stasis this is an easy task. So now, in addition to the framehook logic,
there is a stasis subscription used to determine when the transferee has entered
a bridge. Once it has entered, an appropriate NOTIFY is sent.
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This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk.  In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.

With this the schema only needs to be maintained in one place instead of
once per database.  The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes).  Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.

See contrib/ast-db-manage/README.md for more details.

Review: https://reviewboard.asterisk.org/r/2731

patch by Russell Bryant (license 6300)
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This updates the Asterisk 12 CHANGES file with the things that were missed
during the development cycle.

Review: https://reviewboard.asterisk.org/r/2795/
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