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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Dialog matching is performed in the distributor for the sole
purpose of retrieving an associated serializer so the request
may be serialized.

This patch fixes two problems.

First, incoming CANCEL requests that had no to-tag (which really
should be *all* CANCEL requests) would not match with a dialog.
An earlier bug fix to deal with early CANCEL requests would result
in the CANCEL being replied to with a 481. The fix for this is to
find the matching INVITE transaction and get the dialog from that
transaction.

Second, no SIP responses were matching dialogs. This is because we
were inverting the tags that we were passing into PJSIP's dialog
finding function. This logic has been corrected by setting local
and remote tag variables based on whether the incoming message is
a request or response.
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Stasis events (which get distributed over the ARI WebSocket) are created
by subscribing to the channel_all_cached and bridge_all_cached topics,
filtering out events for channels/bridges currently subscribed to.

There are two issues with that. First was a race condition, where
messages in-flight to the master subscribe-to-all-things topic would get
sent out, even though the events happened before the channel was put
into Stasis. Secondly, as the number of channels and bridges grow in the
system, the work spent filtering messages becomes excessive.

Since r395954, individual channels and bridges have caching topics, and
can be subscribed to individually. This patch takes advantage, so that
channels and bridges are subscribed to on demand, instead of filtering
the global topics.

The one case where filtering is still required is handling BridgeMerge
messages, which are published directly to the bridge_all topic.

Other than the change to how subscriptions work, this patch mostly just
moves code around. Most of the work generating JSON objects from
messages was moved to .to_json handlers on the message types. The
callback functions handling app subscriptions were moved from res_stasis
(b/c they were global to the model) to stasis/app.c (b/c they are local
to the app now).

(closes issue ASTERISK-21969)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2754/
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Storing a backtrace for each allocation in anticipation of a memory
management problem is very CPU intensive.

* Added the CLI "memory backtrace {on|off}" command to request that the
backtrace be gathered only on request.  The backtrace is off by default.

(issue ASTERISK-22221)
Reported by: Matt Jordan
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If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.

This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.

Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.

(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
  issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
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A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.

This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.

Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.

(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
  issueA21064_fix.patch uploaded by wdoekes (License 5674)
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When a channel with the OUTGOING flag leaves a bridge, and it will survive
being pulled from the bridge (either because it will execute dialplan,
go into another bridge, or live in a friendly autoloop), we have to clear
the OUTGOING flag. This is the signal to the CDR engine that this channel
is no longer a second class citizen, i.e., it is not "dialed".

The soft hangup flags are only half the picture. If a channel is being
moved from one bridge to another, the soft hangup flags aren't set; however,
the state of the bridge_channel will not be hung up. Since the channel does
not have one of the two hang up states, that implies that the channel is
still technically alive.

This patch modifies the check so that it checks both the soft hangup flags
as well as the bridge_channel state. If either suggests that the channel
is going to persist, we clear the OUTGOING flag.
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Use a better means of waking up the bridge channel thread.
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After the review for buckets was completed (r2715), the handling of names in
the bucket core was deferred to the wizards. As such, the bucket unit tests
cannot expect that passing a URI with a scheme specified but no actual resource
name will automatically fail. The tests have been updated to not make this
check.
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The config options test requires the entire configuration item to be transparent from
the documentation system. So we let it do that too.

As an aside, please do not use this power for evil. Documentation is your friend, and
you really should document your configurations. Hiding your module's configuration
information from the system attempting to enforce some sanity in the universe is something
only a Bond villain would contemplate.
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The rtpengine configuration parameter was documented in the XML documentation,
but it was not actually registered with the sorcery object. This adds the
parameter with a default of "asterisk", such that res_rtp_asterisk is chosen as
the default RTP implementation.

(closes issue ASTERISK-22380)
Reported by: Rusty Newton
Tested by: Rusty Newton
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Note: This is why I commit on the weekend.


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When originating channels, ast_pbx_outgoing_* caused the dialed channel
reference to be bumped twice. Ostensibly, this routine is bumping the channel
lifetime such that the channel doesn't get nuked in between locks/unlocks;
however, since the routine should return the dialed channel with its
reference bumped, it only needs to do this one time.


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........
Make libuuid an optional dependency for res_rtp_asterisk instead of a requirement.

Review: https://reviewboard.asterisk.org/r/2777/


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Bucket is a URI based API for the creation, retrieval, updating, and deletion
of "buckets" and files contained within them.

Review: https://reviewboard.asterisk.org/r/2715/


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This also adds a _nodoc equivalent.


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(closes issue AST-1200)
reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/2790/



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