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This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
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I've audited all modules that include any header which includes
asterisk/optional_api.h. All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.
In addition ARI dependency declarations have been reworked. Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.
Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606
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* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
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Add bridge_features structure to bridge creation. Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.
ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI
Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
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The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found. The ari stubs though still tried to use the
configuration resulting in segfaults.
This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't. The macro was then added to the mustache
template's "load_module" function.
ASTERISK-27026 #close
Reported-by: Ronald Raikes
Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
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The periodic doc job does a make ari-stubs and checks that
there are no changes before generating the docs. Since I changed
the mustache template (and the generated code directly) recently
and forgot to regenerate the stubs, the doc job thinks they're out
of date.
Change-Id: I94b97035311eccf52b0101b8590223265a7881d4
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The 'ari set debug' command has been enhanced to accept 'all' as an
application name. This allows dumping of all apps even if an app
hasn't registered yet. To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.
'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.
* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
failed, the consumption of the body was moved from the ari stubs
to ast_ari_callback in res_ari.c and the moustache templates were
then regenerated. The body is now passed to ast_ari_invoke and then
on to the handlers. This results in code savings since that template
was inserted multiple times into all the stubs.
An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function. The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.
Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874f39a5a81b20da44358143ed9b54ab0fe)
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In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
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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
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Given resource paths did not have 'json' substituted in for the '{format}'. For
some auto generated documentation/comment strings it resulted in something like
the following:
"... REST handler for /api-docs/sounds.{format}"
This patch makes sure the resource api's path is properly substituted.
ASTERISK-25472 #close
Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23
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Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.
Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.
In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.
It's important to note the following:
- If an offset is provided to the 'play' operations, it only applies to the
first media URI, as it would be weird to skip n seconds forward in every
media resource.
- Operations that control the position of the media only affect the current
media being played. For example, once a media resource in the list
completes, a 'reverse' operation on a subsequent media resource will not
start a previously completed media resource at the appropiate offset.
- This patch does not add any new operations to control the list. Hopefully,
user feedback and/or future patches would add that if people want it.
ASTERISK-26022 #close
Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f
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Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
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ARI modules that are generated by 'make ari-stubs' are all dependent on
res_ari_model. Additionally some of the same modules depend on one or more
res_stasis_* modules.
ASTERISK-25027 #close
Reported by: Corey Farrell
Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153
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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
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This patch improves the documentation of ARI on the wiki. Specifically, it
addresses the following:
* Allowed values and allowed ranges weren't documented. This was particularly
frustrating, as Asterisk would reject query parameters with disallowed values
- but we didn't tell anyone what the allowed values were.
* The /play/id operation on /channels and /bridges failed to document all of
the added media resource types.
* Documentation for creating a channel into a Stasis application failed to
note when it occurred, and that creating a channel into Stasis conflicts with
creating a channel into the dialplan.
* Some other minor tweaks in the mustache templates, including italicizing the
parameter type, putting the default value on its own sub-bullet, and some
other nicities.
Review: https://reviewboard.asterisk.org/r/4351
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Merged revisions 431145 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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"module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/3802
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Previously multiple play actions against a bridge at one time would cause
the sounds to play simultaneously on the bridge. Now if a sound is already
playing, the play action will queue playback to occur after the completion
of other sounds currently on the queue.
(closes issue ASTERISK-22677)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3379/
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Merged revisions 412639 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
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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
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This adds back in support for specifying channel variables during an
originate without compromising the ability to specify query parameters
in the JSON body. This was accomplished by generating the body-parsing
code in a separate function instead of being integrated with the URI
query parameter parsing code such that it could be called by paths with
body parameters. This is transparent to the user of the API and
prevents manual duplication of code or data structures.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3122/
Reported by: Matt Jordan
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Merged revisions 406003 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).
For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.
The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.
(closes issue ASTERISK-22685)
Review: https://reviewboard.asterisk.org/r/2994/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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While working on building client libraries from the Swagger API, I
noticed a problem with the nicknames.
channel.deleteChannel()
channel.answerChannel()
channel.muteChannel()
Etc. We put the object name in the nickname (since we were generating C
code), but it makes OO generators redundant.
This patch makes the nicknames more OO friendly. This resulted in a lot
of name changing within the res_ari_*.so modules, but not much else.
There were a couple of other fixed I made in the process.
* When reversible operations (POST /hold, POST /unhold) were made more
RESTful (POST /hold, DELETE /unhold), the path for the second operation
was left in the API declaration. This worked, but really the two
operations should have been on the same API.
* The POST /unmute operation had still not been REST-ified.
Review: https://reviewboard.asterisk.org/r/2940/
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Merged revisions 402528 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Asterisk will now issue 422 if recording is requested against channels
or bridges with an unknown format
(closes issue ASTERISK-22626)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2939/
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Merged revisions 402001 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If a file already exists in the recordings directory with the same name as what
we would record, issue a 422 instead of relying on the internal failure and
issuing success.
(closes issue ASTERISK-22623)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2922/
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Merged revisions 401973 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/2911/
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Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.
(closes issue ASTERISK-22627)
Reported by: Joshua Colp
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Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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Merged revisions 397816 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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Adds ARI functions to be able to turn on/off music on hold in a
bridge. It actually functions more as a background music without
further actions on the bridge since if the rest of the channels
in the bridge aren't explicitly muted, they will still be able
to communicate.
(closes issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2688/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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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
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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
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