This prevents swap optimization, merges, and transfers involving Stasis
application bridges. It wouldn't be nice if the bridge you thought you
owned disappeared from under you.

Reported-by: Richard Mudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch changes ARI bridging to allow other channel operations to
happen while the channel is bridged.

ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was bridged,
any other channel operations would queue up and execute only after the
channel left the bridge.

This patch changes ARI bridging so that channel commands can execute
while the channel is bridged. For most operations, things simply work
as expected. The one thing that ended up being a bit odd is recording.

The current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself may
be recording; it's recording a specific channel in the bridge that
fails. While this is an annoying limitation, channel recording is
still very useful for use cases such as voice mail, and bridge
recording makes up much of the difference for other use cases.

(closes issue ASTERISK-22084)
Review: https://reviewboard.asterisk.org/r/2726/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These are needed by the pjsip inbound registration test suite tests.

(issue ASTERISK-21833)
(issue ASTERISK-21834)
(issue ASTERISK-21835)
(issue ASTERISK-21837)

Review: https://reviewboard.asterisk.org/r/2700/
Review: https://reviewboard.asterisk.org/r/2739/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396552 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The lonely flag is an optional flag for bridge channels that will
make them leave a bridge when a channel leaves if only lonely
channels are in the bridge at that point. This is useful for things
like ending recording and playback channels when they cease to be
interacting with other channels in the bridge.

(closes issue ASTERISK-22117)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2721/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Expose initial size, automatic increment, maximum size, and idle
timeout as configurable parameters for the res_pjsip thread pool.

Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This crash was caused by decrementing the reference count of a newly created message when
it should not be. This change fixes that but also fixes all other cases where this was
incorrectly done.

(closes issue ASTERISK-22188)
Reported by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds AMI events whenever an outbound registration attempt succeeds
or fails from res_pjsip_outbound_registration. This brings it inline with
the existing SIP channel driver and IAX channel driver.

Review: https://reviewboard.asterisk.org/r/2729/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396201 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.

(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds the following unit tests:
* create_lot: tests adding and removal of a new parking lot (baseline)
* park_extensions: creates a parking lot that registers extensions and
      then confirms that all of the expected extensions exist
* extensions_conflicts: creates numerous parking lots to test that
      extension conflicts in parking lots result in parking lot
      creation failing
* dynamic_parking_variables: Tests that the creation of dynamic
      parking lots respects the related channel variables set on the
      channel that requests them.
* park_call: Tests adding a channel to a parking lot's holding bridge
      by standard parking functions.
* retrieve_call: Tests pulling a channel out of a parking lot's
      holding bridge via parked call retrieval functions.

(closes issue ASTERISK-22138)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2714/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.

This patch renames the variables, adding the ast_ prefix so they will
be exported.

Review: https://reviewboard.asterisk.org/r/2737


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396126 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds basic system information access to ARI.

The results are roughly what you get from 'core show settings', with a
few minor differences.

 * Data is structured, with 'build', 'system', 'config' and 'status'
   sub-objects.
 * Each sub-object is selectable, using the ?only= parameter. A comma
   separated list can be provided to select multiple sections.
 * A few config options are numeric, for which 0 means 'unlimited'.
   Instead of having a special interpretation of those fields, they
   are simply omitted if they're 0.
 * The information is limited to what might be useful to building
   external applications.

(closes issue ASTERISK-21575)
Review: https://reviewboard.asterisk.org/r/2702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396125 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Swagger allows parameters to be specified as 'allowMultiple', meaning
that the parameter may be specified as a comma separated list of
values.

I had written some of the API docs using that, but promptly forgot
about implementing it. This patch finally fills in that gap.

The codegen template was updated to represent 'allowMultiple' fields
as array/size fields in the _args structs. It also parses the comma
separated list using ast_app_separate_args(), so quoted strings in the
argument will be handled properly.

Review: https://reviewboard.asterisk.org/r/2698/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In tracking down some unit tests failures, I ended up reading the fine
print[1] regarding Jansson's thread safety.

In short:
 1. Ref-counting is non-atomic.
 2. json_dumps() and friends are not thread safe.

This patch adds locking where necessary to our ast_json_* wrapper API,
with documentation in json.h describing the thread safety limitations of
the API.

 [1]: http://www.digip.org/jansson/doc/2.4/portability.html#thread-safety

Review: https://reviewboard.asterisk.org/r/2716/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396119 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* BridgeEnter now contains the unique ID of the channel that is to be swapped out, if applicable.
* There is a ParkedCallSwap event that is sent when a parked channel has a new channel take its place.

(closes issue ASTERISK-22193)
reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/2712



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396107 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.

This also fixes two refcounting bugs in the outbound registration code.

Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AST-1198)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * It adds support for externally initiated parking requests. In particular,
   chan_skinny has a protocol level message that initiates a call park.
   This patch now supports that option, as well as the protocol specific
   mechanisms in chan_dahdi/sig_analog and chan_mgcp.
 * A parking bridge features virtual table has been added that provides
   access to the parking functionality that the Bridging API needs. This
   includes requests to park an entire 'call' (with little or no additional
   information, thank you chan_skinny), perform a blind transfer to a parking
   extension, determine if an extension is a parking extension, as well as the
   actual "do the parking" request from the Bridging API.
 * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
   functions
 * The removal of some - but not all - dead parking code from features.c

This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)

Review: https://reviewboard.asterisk.org/r/2710

(closes issue ASTERISK-22134)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Just a merge error due to a file rename. Grrr...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395971 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.

To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.

In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:

     single_topic  ---------------->  all_topic
           ^
           |
     single_topic_cached  ----+---->  all_topic_cached
                              |
                              +---->  cache

This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.

Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.

(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395938 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows starting playback of audio through the CONTROL STREAM FILE
AGI command to start at a particular offset. It will also return the final
position of the file in the 'endpos' attribute.

(closes issue ASTERISK-17803)
Reported by: Murray Melvin
patches:
  res_agi.c.r316293.diff uploaded by murraytm (license 6221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395906 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395884 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was created as a debugging tool before proper endpoint identifiers
were created. Using it now can actually lead to harmful results.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395824 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395779 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22135)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2707



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2692/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395728 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22023)
Reported by Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395674 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The monitor thread is already properly torn down on unload and load
failure.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This renames all files and API calls from several variants of
Stasis-HTTP to ARI including:
* Stasis-HTTP -> ARI
* STASIS_HTTP -> ARI
* stasis_http -> ari (ast_ari for global symbols, file names as well)
* stasis http -> ARI

Review: https://reviewboard.asterisk.org/r/2706/
(closes issue ASTERISK-22136)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most hook callbacks did not need the bridge parameter.  The pointer value
could become invalid if the channel is moved to another bridge while it is
executing.

* Fixed some issues in feature_attended_transfer() as a result.

* Reduce the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the bridge
channel hooks.

* Removed basic bridge requirement on feature_blind_transfer().  It does
not require the basic bridge like feature_attended_transfer().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed feature limits to not use special members of struct
ast_bridge_features.

* Fixed memory leak in off nominal paths of bridge_builtin_set_limits().

* Fixed off nominal path in ast_bridge_features_limits_construct() freeing
unallocated memory if it was not called by bridge_builtin_set_limits().

* Made bridge_builtin_interval_features.so unloadable.

* Simplified parking's use of its duration interval hook.

* Made BridgeWait S option not depend upon another module being loaded.

(closes issue ASTERISK-22107)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2701/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A typo in recent changes caused the JSON ApplicationReplaced message to
fail to build, so the message wasn't being sent out the WebSocket.

Related, the replaced application would also unregister itself when it
disconnected, which would actually unregister the new application. This
was also fixed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395527 65c4cc65-6c06-0410-ace0-fbb531ad65f3