ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
* We weren't properly subscribing to the channel and it's originator
on create.
* We weren't doing a publish_dial after calling ast_call on dial.
* We weren't calling depart_bridge when a channel left the dial bridge.
The first 2 issues were causing events to not be generated and the third
was actually causing channels to not get properly destroyed when hung up.
Together these 3 issues were causing the new
rest_apichannels/create_dial_bridge tests to fail.
As a result of the fixes, the cdr state machine had to be slightly
tweaked to allow bridge leave events without asserting and the tests
themselves had to be updated to account for the channels now cleaning
themselves up.
Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8
ARI dial had been implemented using the Dial API. This made great sense
when dialing was 100% separate from bridging. However, if a channel were
to be added to a bridge during the dial attempt, there would be a
conflict between the dialing thread and the bridging thread. Each would
be attempting to read frames from the dialed channel and act on them.
The initial attempt to make the two play nice was to have the Dial API
suspend the channel in the bridge and stay in charge of the channel
until the dial was complete. The problem with this was that it was
riddled with potential race conditions. It also was not well-suited for
the case where the channel changed which bridge it was in during the
dial.
This new approach removes the use of the Dial API altogether. Instead,
the channel we are dialing is placed into an invisible ARI dialing
bridge. The bridge channel thread handles incoming frames from the
channel. If the channel is added to a real bridge, it is departed from
the invisible bridge and then added to the real bridge. Similarly, if
the channel is removed from the real bridge, it is automatically added
back to the invisible bridge if the dial attempt is still active.
This approach keeps the threading simple by always having the channel
being handled by bridge channel threads.
ASTERISK-25925
Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb
We lose the fact that there is a swap channel if there is one. We
currently wind up rejoining the stasis bridge as a normal join after the
swap channel has already been kicked from the bridge.
This patch preserves the swap channel so the AMI/ARI events can note that
the channel joining the bridge is swapping with another channel. Another
benefit to swaqpping in one operation is if there are any channels that
get lonely (MOH, bridge playback, and bridge record channels). The lonely
channels won't leave before the joining channel has a chance to come back
in under stasis if the swap channel is the only reason the lonely channels
are staying in the bridge.
ASTERISK-25947 #close
Reported by: Richard Mudgett
ASTERISK-24649
Reported by: John Bigelow
ASTERISK-24782
Reported by: John Bigelow
Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee
This adds a new ARI method that allows for you to dial a channel that
you previously created in ARI.
By combining this with the create method for channels, it allows for a
workflow where a channel can be created, manipulated, and then dialed.
The channel is under control of the ARI application during all stages of
the Dial and can even be manipulated based on channel state changes
observed within an ARI application.
The overarching goal for this is to eventually be able to add a dialed
channel to a Stasis bridge earlier than the "Up" state. However, at the
moment more work is needed in the Dial and Bridge APIs in order to
facilitate that.
ASTERISK-25889 #close
Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205
The stasis_app_playback and stasis_app_recording structs need to have a
struct stasis_app_control ref. Other threads can get a reference to the
playback and recording structs from their respective global container.
These other threads can then use the control pointer they contain after
the control struct has gone.
* Add control ref to stasis_app_playback and stasis_app_recording structs.
With the refs added, the control command queue can now have a circular
control reference which will cause the control struct to never get
released if the control's command queue is not flushed when the channel
leaves the Stasis application. Also the command queue needs better
protection from adding commands if the control->is_done flag is set.
* Flush the control command queue on exit.
ASTERISK-25882 #close
Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d
* Give the struct stasis_app_control ao2 object a ref to the channel held
in the object. Now the channel will still be around if a thread needs to
post a stasis message instead of crash because the topic was destroyed.
* Moved stopping any lingering silence generator out of the struct
stasis_app_control destructor and made it a part of exiting the Stasis
application. Who knows which thread the destructor will be called under
so it cannot affect the channel's silence generator. Not only was the
channel unprotected when the silence generator was stopped, stasis may no
longer even control the channel.
ASTERISK-25882
Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4
A recent change to queue channel variable setting to the Stasis control
queue caused a regression. When setting channel variables, it is
possible to give a NULL channel variable value in order to unset the
variable (i.e. remove it from the channel variable list). The change
introduced a call to ast_variable_new(), which is not tolerant of NULL
channel variable values.
This new change switches from using ast_variable to using a custom
channel variable struct that is lighter weight and NULL value-tolerant.
Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d
When queuing tasks onto the Stasis control queue, you can pass an
arbitrary data pointer and a function to free that data. All ARI
commands that use the Stasis control queue made the assumption that the
destructor function would be called in all paths, whether the task was
queued successfully or not. However, this was not correct. If a task was
queued onto a control structure that was already completed, the
allocated data would not be freed properly.
This patch corrects this by making sure that all return paths call the
data destructor.
Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb
A crash occurred when attempting to set a channel variable on a channel
that had already been hung up. This is because there is a small window
between when a control is grabbed and when the channel variable is set
that the channel can be hung up.
The fix here is to queue the setting of the channel variable onto the
control queue. This way, the manipulation of the channel happens in a
thread where it is safe to be done.
In this change, I also noticed that the setting of bridge roles on
channels was being done outside of the control queue, so I also changed
those operations to be done in the control queue.
ASTERISK-25709 #close
Reported by Mark Michelson
Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78
This resolves two observed race conditions.
First, a bit of background on what the Stasis application does:
1a Creates a stasis_app_control structure. This structure is linked into
a global container and can be looked up using a channel's unique ID.
2a Puts the channel in an event loop. The event loop can exit either
because the stasis_app_control structure has been marked done, or
because of some other factor, such as a hangup. In the event loop, the
stasis_app_control determines if any specific ARI commands need to be
run on the channel and will run them from this thread.
3a Checks if the channel is bridged. If the channel is bridged, then
ast_bridge_depart() is called since channels that are added to Stasis
bridges are always imparted as departable.
4a Unlink the stasis_app_control from the container.
When an ARI command is received by Asterisk, the following occurs
1b A thread is spawned to handle the HTTP request
2b The stasis_app_control(s) that corresponds to the channel(s) in the
request is/are retrieved. If the stasis_app_control cannot be
retrieved, then it is assumed that the channel in question has exited
the Stasis app or perhaps was never in Stasis in the first place.
3b A command is queued onto the stasis_app_control, and the channel's
event loop thread is signaled to run the command.
4b While most ARI commands do nothing further, some, such as adding or
removing channels from a bridge, will block until the command they
issued has been completed by the channel's event loop.
The first race condition that is solved by this patch involves a crash
that can occur due to faulty detection of the channel's bridged status
in step 3a. What can happen is that in step 2a, the event loop may run
the ast_bridge_impart() function to asynchronously place the channel
into a bridge, then immediately exit the event loop because the channel
has hung up. In step 3a, we would detect that the channel was not
bridged and would not call ast_bridge_depart(). The reason that the
channel did not appear to be bridged was that the depart_thread that is
spawned by ast_bridge_impart() had not yet started. That is the thread
where the channel is marked as being bridged. Since we did not call
ast_bridge_depart(), the Stasis application would exit, and then the
channel would be destroyed Then the depart_thread would start up and
try to manipulate the destroyed channel, causing a crash.
The fix for this is to switch from using ast_channel_is_bridged() to
checking the NULLity of ast_channel_internal_bridge_channel() to
determine if ast_bridge_depart() needs to be called. The channel's
internal bridge_channel is set when ast_bridge_impart() is called and
is NULLed by the call to ast_bridge_depart(). If the channel's internal
bridge_channel is non-NULL, then the channel must have been imparted
into the bridge and needs to be departed, even if the actual bridging
operation has not yet started. By departing the channel when necessary,
the thread that is running the Stasis application will block until the
bridge gives the okay that the depart_thread has exited.
The second race condition that is solved by this patch involves a leak
of HTTP handler threads. The problem was that step 2b would successfully
retrieve a stasis_app_control structure. Then step 2a would exit the
channel from the event loop due to a hangup. Steps 3a and 4a would
execute, and then finally steps 3b and 4b would. The problem is that at
step 4b, when attempting to add a channel to a bridge, the thread would
block forever since the channel would never execute the queued command
since it was finished with the event loop. This meant that the HTTP
handling thread would be leaked, along with any references that thread
may have owned (in my case, I was seeing bridges leaked).
The fix for this is to hone in better on when the channel has exited the
event loop. The stasis_app_control structure has an is_done field that
is now set at each point where the channel may exit the event loop. If
step 2b retrieves a valid stasis_app_control structure but the control
is marked as done, then the attempted operation exits immediately since
there will be nothing to service the attempted command.
ASTERISK-25091 #close
Reported by Ilya Trikoz
Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
app_control_register_rule and app_control_unregister_rule lock/unlock
the queue, which is a mutating operation according to the
ao2_lock/_unlock prototype. Depending on the specific (implicit) casts
in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only
callers of those functions do not have the const, get consistent results
by just dropping it.
Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
only transfer channels to a SIP URI, i.e., you had to pass
'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
still supported, it is somewhat unintuitive - particularly in a world full
of endpoints. As such, we now also support specifying the PJSIP endpoint to
transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
updating its Contact header. Alas, that resulted in the forwarding
destination set by the dialplan application/ARI resource/whatever being
rewritten with very incorrect information. Hence, we now don't bother
updating an outgoing response if it is a 302. Since this took a looong time
to find, some additional debug statements have been added to those modules
that update the Contact headers.
Review: https://reviewboard.asterisk.org/r/4316/
ASTERISK-24015 #close
Reported by: Private Name
ASTERISK-24703 #close
Reported by: Matt Jordan
........
Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses a few issues:
1) The order of Dial events have been changed when performing a call forward.
The order has now been altered to
1) Dial begins dialing channel A.
2) When A forwards the call to B, we issue the dial end event to channel
A, indicating the dial is being canceled due to a forward to B.
3) When the call to channel B occurs, we then issue a new dial begin to
channel B.
2) Call forwards are now reported on the calling channel, not the peer channel.
3) AMI DialEnd events have been altered to display the extension the call is
being forwarded to when relevant.
4) You can now get the values of channel variables for channels that are not
currently in the Stasis application. This brings the retrieval of channel
variables more in line with the rest of channel read operations since they
may be performed on channels not in Stasis.
ASTERISK-24134 #close
Reported by Matt Jordan
ASTERISK-24138 #close
Reported by Matt Jordan
Patches:
forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
Review: https://reviewboard.asterisk.org/r/3899
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Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes an issue where a Stasis application running over ARI and
subscribed to ari/events could miss the ChannelEnteredBridge event
because it did not subscribe to the new bridge fast enough.
To accomplish this, it subscribes the application controlling the
channel to the new bridge before adding it to that bridge which
required the stasis_app_control structure to maintain a reference to
the stasis_app.
(closes issue ASTERISK-23295)
Review: https://reviewboard.asterisk.org/r/3336/
........
Merged revisions 410527 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
........
Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge. In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.
Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.
(closes issue ASTERISK-22624)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2947/
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Merged revisions 403749 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
........
Merged revisions 403342 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).
Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.
(closes issue ASTERISK-22701)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2916/
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Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.
(closes issue ASTERISK-22703)
Reported by: John Bigelow
(closes issue ASTERISK-22634)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2965/
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Merged revisions 402336 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
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Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds error checking to ARI bridge operations, when
adding/removing channels to/from bridges.
In general, the error codes fall out as follows:
* Bridge not found - 404 Not Found
* Bridge not in Stasis - 409 Conflict
* Channel not found - 400 Bad Request
* Channel not in Stasis - 422 Unprocessable Entity
* Channel not in this bridge (on remove) - 422 Unprocessable Entity
(closes issue ASTERISK-22036)
Review: https://reviewboard.asterisk.org/r/2769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 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
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
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
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
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds support for Stasis applications to receive bridge-related
messages when the application shows interest in a given bridge.
To supplement this work and test it, this also adds support for the
following bridge-related Stasis-HTTP functionality:
* GET stasis/bridges
* GET stasis/bridges/{bridgeId}
* POST stasis/bridges
* DELETE stasis/bridges/{bridgeId}
* POST stasis/bridges/{bridgeId}/addChannel
* POST stasis/bridges/{bridgeId}/removeChannel
Review: https://reviewboard.asterisk.org/r/2572/
(closes issue ASTERISK-21711)
(closes issue ASTERISK-21621)
(closes issue ASTERISK-21622)
(closes issue ASTERISK-21623)
(closes issue ASTERISK-21624)
(closes issue ASTERISK-21625)
(closes issue ASTERISK-21626)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk REST interface.
This adds the /playback/{playbackId}/control resource, which may be
POSTed to to pause, unpause, reverse, forward or restart the media
playback.
Attempts to control a playback that is not currently playing will
either return a 404 Not Found (because the playback object no longer
exists) or a 409 Conflict (because the playback object is still in the
queue to be played).
This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource.
(closes issue ASTERISK-21587)
Review: https://reviewboard.asterisk.org/r/2559
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and GET /playback/{playbackId}.
This allows an external application to initiate playback of a sound on a
channel while the channel is in the Stasis application.
/play commands are issued asynchronously, and return immediately with
the URL of the associated /playback resource. Playback commands queue up,
playing in succession. The /playback resource shows the state of a
playback operation as enqueued, playing or complete. (Although the
operation will only be in the 'complete' state for a very short time,
since it is almost immediately freed up).
(closes issue ASTERISK-21283)
(closes issue ASTERISK-21586)
Review: https://reviewboard.asterisk.org/r/2531/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.
This patch breaks the major components of res_stasis.c into individual
files.
* res/stasis/app.c - Stasis application tracking
* res/stasis/control.c - Channel control objects
* res/stasis/command.c - Channel command object
This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.
The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.
Review: https://reviewboard.asterisk.org/r/2530/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
