In Asterisk, it is possible for a device to have a status of ONHOLD. This is
not typically an easy thing to determine, as a channel being on hold is not
a direct channel state. Typically, this has to be calculated outside of the
core independently in channel drivers, notably, chan_sip and chan_pjsip. Both
of these channel drivers already have to calculate device state in a fashion
more complex than the core can handle, as they aggregate all state of all
channels associated with a peer/endpoint; they also independently track
whether or not one of those channels is currently on hold and mark the device
state appropriately.

In 12+, we now have the ability to report an AST_DEVICE_ONHOLD state for all
channels that defer their device state to the core. This is due to channel hold
state actually now being tracked on the channel itself. If a channel driver
defers its device state to the core (which many, such as DAHDI, IAX2, and
others do in most situations), the device state core already goes out to get a
channel associated with the device. As such, it can now also factor the channel
hold state in its calculation.

This patch adds this logic to the device state core. It also uses an existing
mapping between device state and channel state to handle more channel states.
chan_pjsip has been updated slightly as well to make use of this (as it was,
for some reason, reporting a channel state of BUSY as a device state of INUSE,
which feels slightly wrong).

Review: https://reviewboard.asterisk.org/r/3771/

ASTERISK-24038 #close


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.

Review: https://reviewboard.asterisk.org/r/3812/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does two things:
(1) It updates the unit tests to expect additional stasis messages. More
    messages are now sent to the endpoint topic, due to forwarding all
    channel messages and the forwarding relationship set up between
    endpoints themselves.
(2) Remove the technology forwarding subscription during
    ast_endpoint_shutdown. This prevents an improper double shutdown of
    an endpoint from occurring.
........

Merged revisions 419318 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
res_pjsip_refer: remove stray debugging line

How'd those @ symbols get in there...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When updating voicemail.conf when a user changes
their pin, change the generator string to be the
same as the module name when reading so that the
same config_hook will be called.

Review: https://reviewboard.asterisk.org/r/3837/
........

Merged revisions 419284 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 419285 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419286 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Unregister manager actions FAXSessions, FAXSession and FAXStats at unload.
* Update ast_manager_register2 use ao2_t_alloc tagged with the action name.

ASTERISK-24058 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3831/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Say you wanted to include variables in an application map and have those
variables substituted and passed along to the application being executed;
currently this does not happen.

This patch adds this ability to pass channel variable values to an
application before being executed.

ASTERISK-22608 #close
Reported by: Michael L. Young
patches:
  features_substitute_arguments_v2.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3819/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419252 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We have a new periodic beep feature but sometimes a user needs some sort of
feedback, without the need to have a periodic beep during the recording, to let
them know that MixMonitor started recording or ended the recording.  The use
case where this patch is being used is when using Dynamic Features to start and
end MixMonitor.

This patch adds an option to play a beep when MixMonitor starts and an option to
play a beep when MixMonitor ends.

ASTERISK-24051 #close
Reported by: Michael L. Young
patches:
  mixmonitor-play-beep-start-stop.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3820/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When updating a row, we are currently doing an INSERT OR REPLACE INTO.  The
downside to this is that the row is deleted if it exists and then a new row is
inserted.  So, we are hitting the disk twice.  One for the deletion and one for
the insertion.

This patch changes this statement to an INSERT INTO and if the insert fails
because a row with that key exists, we will IGNORE the failure.  Then we will
attempt to perform an UPDATE on the existing row if that row wasn't just
INSERTed.

ASTERISK-24050 #close
Reported by: Michael L. Young
patches:
  astdb-insert-update-io-help_trunk_v2.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3815/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419222 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Made use a local static frame to generate the AST_FRAME_CNG frame when
silence starts.

I don't think the handling of the AST_FRAME_CNG has ever really worked
because there doesn't seem to be any consumers of it.

Review: https://reviewboard.asterisk.org/r/3813/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419206 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
    channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
    for sending/receiving arbitrary out of call text messages through ARI in a
    technology agnostic fashion.

The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
    relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
    arbitrary technology defined URI. This is less straight forward, as
    endpoints are formed from a tech + resource pair. We don't have a
    mechanism to note that a technology that *may* have endpoints exists.

This patch provides such a mechanism, and fixes a few bugs along the way.

The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
    most of the interesting bits (such as channel creation, destruction, state
    changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
    This resulted in endpoints missing the channel creation message, which
    limited the usefulness of the subscription in the first place (a major use
    case being 'tell me when this endpoint has a channel'). Unfortunately,
    this meant another parameter to ast_channel_alloc. Since not all channel
    technologies support an ast_endpoint, this patch makes such a call
    optional and opts for a new function, ast_channel_alloc_with_endpoint.

When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.

Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:

channel PJSIP/foo-1 --
                      \
                       --> endpoint PJSIP/foo --
                      /                         \
channel PJSIP/foo-2 --                           \
                                                  ---- > endpoint PJSIP
                                                /
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --

ARI, through the applications resource, can:
 - subscribe to endpoint:PJSIP/foo and get notifications for channels
   PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
 - subscribe to endpoint:PJSIP/bar and get notifications for channels
   PJSIP/bar-1 and endpoint PJSIP/bar
 - subscribe to endpoint:PJSIP and get notifications for channels
   PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar

Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).

This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).

Review: https://reviewboard.asterisk.org/r/3760/

ASTERISK-23692
........

Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The iax2_best_codec function was changed to convert the formats
into a format compatibilities structure and grab the first
format from it. The resulting order differs from the previous
order of iax2_best_codec which causes unexpected formats to
get chosen (such as g723).

This commit brings back the old behavior of iax2_best_codec by
having a specified preference list.

Review: https://reviewboard.asterisk.org/r/3835/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419180 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 419162 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 419163 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents a crash in the Dial API triggered by use of the Page()
application where a format capability struct was used before checking
whether it was NULL.

ASTERISK-24074 #close


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419111 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After merging the media formats branch, chan_iax2 was discarding
codec preferences for the purpose of choosing which codec a
channel would use once a call started. This patch restores the
Asterisk 1.8-12 codec choice behaviors.

ASTERISK-23958 #close
Review: https://reviewboard.asterisk.org/r/3800/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-24072 #close
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419093 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When configuring Asterisk to build against a version of pjproject installed
in a non-standard location, the checks for "PJSIP Transaction Group Lock
Support" and "PJSIP Media Stream Replacement Support" fail.  This is
because these secondary checks are not taking the CFLAGS and LIBS returned
by the pkg-config check into account.

Review: https://reviewboard.asterisk.org/r/3830


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419077 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Remove functions:
	ast_smdi_interface_unref
	ast_smdi_md_message_putback
	ast_smdi_mwi_message_putback
	ast_smdi_md_message destructor
	ast_smdi_mwi_message destructor

Includes for astobj.h are removed everywhere it's possible.

ASTERISK-24066 #close
Review: https://reviewboard.asterisk.org/r/3758/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
 1. Asterisk was limited in how many formats it could handle.
 2. Formats, being a bit field, could not include any attribute information.
    A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.

Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.

Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.

Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.

Generally, the new philosophy for handling formats is as follows:
 * The ast_format structure is reference counted. This removed a large amount
   of the memory allocations and copying that was done in prior versions.
 * In order to prevent race conditions while keeping things performant, the
   ast_format structure is immutable by convention and lock-free. Violate this
   tenet at your peril!
 * Because formats are reference counted, codecs are also reference counted.
   The Asterisk core generally provides built-in codecs and caches the
   ast_format structures created to represent them. Generally, to prevent
   inordinate amounts of module reference bumping, codecs and formats can be
   added at run-time but cannot be removed.
 * All compatibility with the bit field representation of codecs/formats has
   been moved to a compatibility API. The primary user of this representation
   is chan_iax2, which must continue to maintain its bit-field usage of formats
   for interoperability concerns.
 * When a format is negotiated with attributes, or when a format cannot be
   represented by one of the cached formats, a new format object is created or
   cloned from an existing format. That format may have the same codec
   underlying it, but is a different format than a version of the format with
   different attributes or without attributes.
 * While formats are reference counted objects, the reference count maintained
   on the format should be manipulated with care. Formats are generally cached
   and will persist for the lifetime of Asterisk and do not explicitly need
   to have their lifetime modified. An exception to this is when the user of a
   format does not know where the format came from *and* the user may outlive
   the provider of the format. This occurs, for example, when a format is read
   from a channel: the channel may have a format with attributes (hence,
   non-cached) and the user of the format may last longer than the channel (if
   the reference to the channel is released prior to the format's reference).

For more information on this work, see the API design notes:
  https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite

Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.

There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).

Reviews:
 https://reviewboard.asterisk.org/r/3814
 https://reviewboard.asterisk.org/r/3808
 https://reviewboard.asterisk.org/r/3805
 https://reviewboard.asterisk.org/r/3803
 https://reviewboard.asterisk.org/r/3801
 https://reviewboard.asterisk.org/r/3798
 https://reviewboard.asterisk.org/r/3800
 https://reviewboard.asterisk.org/r/3794
 https://reviewboard.asterisk.org/r/3793
 https://reviewboard.asterisk.org/r/3792
 https://reviewboard.asterisk.org/r/3791
 https://reviewboard.asterisk.org/r/3790
 https://reviewboard.asterisk.org/r/3789
 https://reviewboard.asterisk.org/r/3788
 https://reviewboard.asterisk.org/r/3787
 https://reviewboard.asterisk.org/r/3786
 https://reviewboard.asterisk.org/r/3784
 https://reviewboard.asterisk.org/r/3783
 https://reviewboard.asterisk.org/r/3778
 https://reviewboard.asterisk.org/r/3774
 https://reviewboard.asterisk.org/r/3775
 https://reviewboard.asterisk.org/r/3772
 https://reviewboard.asterisk.org/r/3761
 https://reviewboard.asterisk.org/r/3754
 https://reviewboard.asterisk.org/r/3753
 https://reviewboard.asterisk.org/r/3751
 https://reviewboard.asterisk.org/r/3750
 https://reviewboard.asterisk.org/r/3748
 https://reviewboard.asterisk.org/r/3747
 https://reviewboard.asterisk.org/r/3746
 https://reviewboard.asterisk.org/r/3742
 https://reviewboard.asterisk.org/r/3740
 https://reviewboard.asterisk.org/r/3739
 https://reviewboard.asterisk.org/r/3738
 https://reviewboard.asterisk.org/r/3737
 https://reviewboard.asterisk.org/r/3736
 https://reviewboard.asterisk.org/r/3734
 https://reviewboard.asterisk.org/r/3722
 https://reviewboard.asterisk.org/r/3713
 https://reviewboard.asterisk.org/r/3703
 https://reviewboard.asterisk.org/r/3689
 https://reviewboard.asterisk.org/r/3687
 https://reviewboard.asterisk.org/r/3674
 https://reviewboard.asterisk.org/r/3671
 https://reviewboard.asterisk.org/r/3667
 https://reviewboard.asterisk.org/r/3665
 https://reviewboard.asterisk.org/r/3625
 https://reviewboard.asterisk.org/r/3602
 https://reviewboard.asterisk.org/r/3519
 https://reviewboard.asterisk.org/r/3518
 https://reviewboard.asterisk.org/r/3516
 https://reviewboard.asterisk.org/r/3515
 https://reviewboard.asterisk.org/r/3512
 https://reviewboard.asterisk.org/r/3506
 https://reviewboard.asterisk.org/r/3413
 https://reviewboard.asterisk.org/r/3410
 https://reviewboard.asterisk.org/r/3387
 https://reviewboard.asterisk.org/r/3388
 https://reviewboard.asterisk.org/r/3389
 https://reviewboard.asterisk.org/r/3390
 https://reviewboard.asterisk.org/r/3321
 https://reviewboard.asterisk.org/r/3320
 https://reviewboard.asterisk.org/r/3319
 https://reviewboard.asterisk.org/r/3318
 https://reviewboard.asterisk.org/r/3266
 https://reviewboard.asterisk.org/r/3265
 https://reviewboard.asterisk.org/r/3234
 https://reviewboard.asterisk.org/r/3178

ASTERISK-23114 #close
Reported by: mjordan
  media_formats_translation_core.diff uploaded by kharwell (License 6464)
  rb3506.diff uploaded by mjordan (License 6283)
  media_format_app_file.diff uploaded by kharwell (License 6464) 
  misc-2.diff uploaded by file (License 5000)
  chan_mild-3.diff uploaded by file (License 5000) 
  chan_obscure.diff uploaded by file (License 5000) 
  jingle.diff uploaded by file (License 5000) 
  funcs.diff uploaded by file (License 5000) 
  formats.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  bridges.diff uploaded by file (License 5000) 
  mf-codecs-2.diff uploaded by file (License 5000) 
  mf-app_fax.diff uploaded by file (License 5000) 
  mf-apps-3.diff uploaded by file (License 5000) 
  media-formats-3.diff uploaded by file (License 5000) 

ASTERISK-23715
  rb3713.patch uploaded by coreyfarrell (License 5909)
  rb3689.patch uploaded by mjordan (License 6283)
  
ASTERISK-23957
  rb3722.patch uploaded by mjordan (License 6283) 
  mf-attributes-3.diff uploaded by file (License 5000) 

ASTERISK-23958
Tested by: jrose
  rb3822.patch uploaded by coreyfarrell (License 5909) 
  rb3800.patch uploaded by jrose (License 6182)
  chan_sip.diff uploaded by mjordan (License 6283) 
  rb3747.patch uploaded by jrose (License 6182)

ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
  sip_cleanup.diff uploaded by opticron (License 6273)
  chan_sip_caps.diff uploaded by mjordan (License 6283) 
  rb3751.patch uploaded by coreyfarrell (License 5909) 
  chan_sip-3.diff uploaded by file (License 5000) 

ASTERISK-23960 #close
Tested by: opticron
  direct_media.diff uploaded by opticron (License 6273) 
  pjsip-direct-media.diff uploaded by file (License 5000) 
  format_cap_remove.diff uploaded by opticron (License 6273) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  chan_pjsip-2.diff uploaded by file (License 5000) 

ASTERISK-23966 #close
Tested by: rmudgett
  rb3803.patch uploaded by rmudgetti (License 5621)
  chan_dahdi.diff uploaded by file (License 5000) 
  
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
  rb3814.patch uploaded by rmudgett (License 5621) 
  moh_cleanup.diff uploaded by opticron (License 6273) 
  bridge_leak.diff uploaded by opticron (License 6273) 
  translate.diff uploaded by file (License 5000) 
  rb3795.patch uploaded by rmudgett (License 5621) 
  tls_fix.diff uploaded by mjordan (License 6283) 
  fax-mf-fix-2.diff uploaded by file (License 5000) 
  rtp_transfer_stuff uploaded by mjordan (License 6283) 
  rb3787.patch uploaded by rmudgett (License 5621) 
  media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) 
  format_cache_case_fix.diff uploaded by opticron (License 6273) 
  rb3774.patch uploaded by rmudgett (License 5621) 
  rb3775.patch uploaded by rmudgett (License 5621) 
  rtp_engine_fix.diff uploaded by opticron (License 6273) 
  rtp_crash_fix.diff uploaded by opticron (License 6273) 
  rb3753.patch uploaded by mjordan (License 6283) 
  rb3750.patch uploaded by mjordan (License 6283) 
  rb3748.patch uploaded by rmudgett (License 5621) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  rb3740.patch uploaded by mjordan (License 6283) 
  rb3739.patch uploaded by mjordan (License 6283) 
  rb3734.patch uploaded by mjordan (License 6283) 
  rb3689.patch uploaded by mjordan (License 6283) 
  rb3674.patch uploaded by coreyfarrell (License 5909) 
  rb3671.patch uploaded by coreyfarrell (License 5909) 
  rb3667.patch uploaded by coreyfarrell (License 5909) 
  rb3665.patch uploaded by mjordan (License 6283) 
  rb3625.patch uploaded by coreyfarrell (License 5909) 
  rb3602.patch uploaded by coreyfarrell (License 5909) 
  format_compatibility-2.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new operation for stored recordings, copy. It takes an
existing stored recording and makes a copy of it in the same directory
or a relative directory under the stored recording directory.

/ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}

This is particularly useful for voicemail-esque applications, which may need to
copy or move recordings around a directory structure.

Review: https://reviewboard.asterisk.org/r/3768/

ASTERISK-24036 #close
Reported by: Sam Galarneau
Tested by: Sam Galarneau
........

Merged revisions 419021 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a build failure introduced by r3821.  struct stasis_topic is
opaque, so topic->name is unavailable.  Switch to using stasis_topic_name().
........

Merged revisions 419019 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add tags to stasis objects using the name.  This makes it
easier to track the source of certain stasis ref leaks.

Review: https://reviewboard.asterisk.org/r/3821/
........

Merged revisions 418996 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418997 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418980 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a reference count goes negative, instead of
just logging that fact, be more helpful with a
backtrace and an assert that will DO_CRASH.

This patch also removes the duplicate ao2_bt()
function and cleans up extraneous usage of the
ast_log_backtrace() call.

Review: https://reviewboard.asterisk.org/r/3765/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Holding a reference to the peer during mwi subscriptions
resulted in a circular reference because the final event
message would not be sent until destruction of the peer.

Instead, pass the name of the peer to the event callback
so that it can fail gracefully after the peer has gone.

ASTERISK-23959
Review: https://reviewboard.asterisk.org/r/3754/
........

Merged revisions 418636 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the features.conf is missing, the cfg->featurgroups
and cfg->applicationmaps is not initialized, resulting
in assert on ao2_find of a null container.  This patch
changes the initialization call and adds asserts for a
safeguard.

Review: https://reviewboard.asterisk.org/r/3809/
........

Merged revisions 418886 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418961 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 418937 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418938 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.

Review: https://reviewboard.asterisk.org/r/3721/
........

Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Specifically the following equivalents were created:
fax show session -> FAXSession
fax show sessions -> FAXSessions
fax show stats -> FAXStats

Review: https://reviewboard.asterisk.org/r/3666/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.

This patch changes the behaviour of this setting in the following ways:

1) In sip_write(), do not pass the media unless we have either progressed
   beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
   media is both set-up and wanted. This helps resolve double-ringing on some
   buggy handsets.

2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
   SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
   enabling early media. Avoid sending double ring indications.

NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.

Review: https://reviewboard.asterisk.org/r/3700

ASTERISK-23972 #close
Reported by: Steve Davies
patches:
  inband_never_present_early_media2 uploaded by Steve Davies (License 5012)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The commit that added libxml2 support didn't fully check for the libxml2
development script in the Asterisk configure file. As a result, Asterisk could
be configured, then fail on menuselect. This patch fixes it so that Asterisk
should detect the libxml2 dependency failure first.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the final patch in adding menuselect to Asterisk.
 - The first patch (r418832) added menuselect along with mxml
 - The second patch (r418833) removed mxml from menuselect

This patch adds support for libxml2 to menuselect, and makes libxml2 a
required library for Asterisk.

Note that the libxml2 portion of this patch was written by Sean Bright,
and was made available on a team branch:
  http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/

Review: https://reviewboard.asterisk.org/r/3773/

ASTERISK-20703 #close
patches:
  some_mysterious_team_branch uploaded by seanbright (License 5060)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the second patch that adds menuselect to Asterisk trunk. The previous
commit (r418832) added menuselect along with mxml; this patch removes mxml
completely from Menuselect.

A subsequent patch will switch menuselect over to using libxml2, and make
libxml2 a required dependency for Asterisk.

ASTERISK-20703


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the first patch that adds menuselect to Asterisk trunk, and removes
the svn:externals property. This is being done for two reasons:
(1) The removal of external repositories eases a future migration to git
(2) Asterisk is now the only thing that uses menuselect; as a result, there's
    little need to keep it in an external repository

Subsequent patches will remove the mxml dependency from menuselect and tidy
up the build system.

ASTERISK-20703


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418832 65c4cc65-6c06-0410-ace0-fbb531ad65f3