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 1d890874f3)
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
This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
ASTERISK-26488 #close
Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
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
This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.
ASTERISK-26470 #close
Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d
Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system. In this case string values
from a channel driver's peer and not from the user setting channel
variables.
* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
* 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
The JSON library Asterisk uses, jansson, is not thread
safe for us in a few ways. To help with this wrappers for JSON
object reference count increasing and decreasing were added
which use a global lock to ensure they don't clobber over
each other. This does not extend to reference count manipulation
within the jansson library itself. This means you can't safely
use the object borrowing specifier (O) in ast_json_pack and
you can't share JSON instances between objects.
This change removes uses of the O specifier and replaces them
with the o specifier and an explicit ast_json_ref. Some cases
of instance sharing have also been removed.
ASTERISK-25601 #close
Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1
This patch adds the ability to subscribe to all events. There are two possible
ways to accomplish this:
(1) On initial WebSocket connection. This patch adds a new query parameter,
'subscribeAll'. If present and True, Asterisk will subscribe the
applications to all ARI events.
(2) Via the applications resource. When subscribing in this manner, an ARI
client should merely specify a blank resource name, i.e., 'channels:'
instead of 'channels:12354'. This will subscribe the application to all
resources of the 'channels' type.
ASTERISK-24870 #close
Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6
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
Prior to this patch, when a WebSocket connection is made, ARI would not
be informed of the connection until after the WebSocket layer had
accepted the connection. This created a brief race condition where the
ARI client would be notified that it was connected, a channel would be
sent into the Stasis dialplan application, but ARI would not yet have
registered the Stasis application presented in the HTTP request that
established the WebSocket.
This patch resolves this issue by doing the following:
* When a WebSocket attempt is made, a callback is made into the ARI
application layer, which verifies and registers the apps presented in
the HTTP request. Because we do not yet have a WebSocket, we cannot
have an event session for the corresponding applications. Some
defensive checks were thus added to make the application objects
tolerant to a NULL event session.
* When a WebSocket connection is made, the registered application is
updated with the newly created event session that wraps the WebSocket
connection.
ASTERISK-24988 #close
Reported by: Joshua Colp
Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636
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
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When swapping a Local channel in place of one already
in a bridge (to complete a bridge attended transfer),
the channel that was swapped out can actually be hung
up before the stasis bridge push callback executes on
the independant transfer thread. This results in the
stasis app loop dropping out and removing the control
that has the the app name which the local replacement
channel needs so it can re-enter stasis.
To avoid this race condition a new push_peek callback
has been added, and called from the ast_bridge_impart
thread before it launches the independant thread that
will complete the transfer. Now the stasis push_peek
callback can copy the stasis app name before the swap
channel can hang up.
ASTERISK-24649
Review: https://reviewboard.asterisk.org/r/4382/
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After a bridge transfer completes where a local replacement
channel is used, a stasis transfer message with the details
of the transfer is sent. This is processed by stasis which
then sets the stasis app name and replaced channel snapshot
on the replacement channel.
However, since a separate thread was already started to run
stasis on the new replacement channel, a race was on to see
if the message processing would be completed before the app
name was needed, otherwise the channel would be hung up.
This change moves the calls used to set the stasis app name
and the replace snapshot to the bridge_stasis_push function
callback from the bridge transfer logic, allowing the steps
to be completed earlier and more deterministically, and the
race elimianted.
NOTE: the swap channel parameter to bridge_stasis_push (and
thus all bridge push callbacks) must always be present when
performing a swap with another channel.
ASTERISK-24649 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4341/
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This corrects several bugs that currently exist in the stasis
application code.
* After a masquerade, the resulting channels have channel topics that
do not match their uniqueids
** Masquerades now swap channel topics appropriately
* StasisStart and StasisEnd messages are leaked to observer
applications due to being published on channel topics
** StasisStart and StasisEnd publishing is now properly restricted
to controlling apps via app topics
* Race conditions exist where StasisStart and StasisEnd messages due to
a masquerade may be received out of order due to being published on
different topics
** These messages are now published directly on the app topic so this
is now a non-issue
* StasisEnds are sometimes missing when sent due to masquerades and
bridge swaps into and out of Stasis()
** This was due to StasisEnd processing adjusting message-sent flags
after Stasis() had already exited and Stasis() had been re-entered
** This was corrected by adjusting these flags prior to sending the
message while the initial Stasis() application was still shutting
down
Review: https://reviewboard.asterisk.org/r/4213/
ASTERISK-24537 #close
Reported by: Matt DiMeo
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From reviewboard:
"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?
The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."
The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.
Review: https://reviewboard.asterisk.org/r/4135
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This change corrects message ordering in cases where a channel-related
message can be received after a Stasis/ARI application has received the
StasisEnd message. The StasisEnd message was being passed to
applications directly without waiting for the channel topic to empty.
As a result of this fix, other bugs were also identified and fixed:
* StasisStart messages were also being sent directly to apps and are
now routed through the stasis message bus properly
* Masquerade monitor datastores were being removed at the incorrect
time in some cases and were causing StasisEnd messages to not be sent
* General refactoring where necessary for the above
* Unsubscription on StasisEnd timing changes to prevent additional
messages from following the StasisEnd when they shouldn't
A channel sanitization function pointer was added to reduce processing
and AO2 lookups.
Review: https://reviewboard.asterisk.org/r/4163/
ASTERISK-24501 #close
Reported by: Matt Jordan
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When ARI manipulates a bridge, it generally doesn't care what the mixing
technology is. Operations on a bridge initiated through ARI should perform
their action in generally the same way, regardless of the bridge's mixing
technology. While the mixing technology may determine how media flows to
channels, the actual operations on a bridge themselves should be the same.
Currently, this isn't the case with holding bridges. When a channel joins
without a role, MoH is started on that channel automatically. Subsequent bridge
operations that would stop MoH would fail (as there is no Announcer channel
playing MoH to the bridge). Starting MoH on the bridge will also create two
MoH streams: one from the MoH being played on the participant channel, and one
from the announcer channel. From the perspective of ARI users, this is
counter-intuitive - I would not expect MoH to be started for me. The mixing
technology determines how media is shared between participants, not the
application experience.
This patch does the following:
* The Stasis bridge class now inspects channels as they are going into a
bridge. If the bridge has a holding capability, and the channel has no
roles, we give it a participant role and mark the default behaviour to have
no entertainment. This allows addChannel operations to continue to set a
participant role with an entertainment option if it felt like it (or could
do it).
* The music on hold channel is now Stasis approved (tm)
Review: https://reviewboard.asterisk.org/r/3929/
ASTERISK-24264 #close
Reported by: Samuel Galarneau
Tested by: Samuel Galarneau
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When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.
This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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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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r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
ARI: Add channel technology agnostic out of call text messaging
This patch adds the ability to send and receive text messages from various
technology stacks in Asterisk through ARI. This includes chan_sip (sip),
res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
endpoints resource, and can be sent directly through that resource, or to a
particular endpoint.
For example, the following would send the message "Hello there" to PJSIP
endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
This is equivalent to the following as well:
ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
Both forms are available for message technologies that allow for arbitrary
destinations, such as chan_sip.
Inbound messages can now be received over ARI as well. An ARI application that
subscribes to endpoints will receive messages from those endpoints:
{
"type": "TextMessageReceived",
"timestamp": "2014-07-12T22:53:13.494-0500",
"endpoint": {
"technology": "PJSIP",
"resource": "alice",
"state": "online",
"channel_ids": []
},
"message": {
"from": "\"alice\" <sip:alice@127.0.0.1>",
"to": "pjsip:asterisk@127.0.0.1",
"body": "Watson, come here.",
"variables": []
},
"application": "testsuite"
}
The above was made possible due to some rather major changes in the message
core. This includes (but is not limited to):
- Users of the message API can now register message handlers. A handler has
two callbacks: one to determine if the handler has a destination for the
message, and another to handle it.
- All dialplan functionality of handling a message was moved into a message
handler provided by the message API.
- Messages can now have the technology/endpoint associated with them.
Various other properties are also now more easily accessible.
- A number of ao2 containers that weren't really needed were replaced with
vectors. Iteration over ao2_containers is expensive and pointless when
the lifetime of things is well defined and the number of things is very
small.
res_stasis now has a new file that makes up its structure, messaging. The
messaging functionality implements a message handler, and passes received
messages that match an interested endpoint over to the app for processing.
Note that inadvertently while testing this, I reproduced ASTERISK-23969.
res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
fix for that as well.
Review: https://reviewboard.asterisk.org/r/3726
ASTERISK-23692 #close
Reported by: Matt Jordan
ASTERISK-23969 #close
Reported by: Andrew Nagy
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r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
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r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
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* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.
AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3720/
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This patch fixes two bugs:
1. When originating a channel into a Stasis application, we already create a
subscription for the channel that is going into our Stasis app.
Unfortunately, when you create a Local channel and pass it off to a Stasis
app, you really aren't creating just one channel: you're creating two. This
patch snags the second half of the Local channel pair (assuming it is a
Local channel pair, but luckily core_local is kind about such assumptions)
and subscribes to it as well.
2. Subscriptions are a bit sticky right now. If a subscription is made, the
'interest' count gets bumped on the Stasis subscription - but unless
something explicitly unsubscribes the channel, said subscription sticks
around. This is not much of a problem is a user is creating the subscription
- if they made it, they must want it. However, when we are creating
implicit subscriptions, we need to make sure something clears them out.
This patch takes a pessimistic approach: it watches the cache updates
coming from Stasis and, if we notice that the cache just cleared out an
object, we delete our subscription object. This keeps our ao2 container of
Stasis forwards in an application from growing out of hand; it also is a
bit more forgiving for end users who may not realize they were supposed to
unsubscribe from that channel that just hung up.
Review: https://reviewboard.asterisk.org/r/3710/
#ASTERISK-23939 #close
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User events can now be generated from ARI. Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots. An application must be specified which will receive
the event message (other applications can subscribe to it). The message
will also be delivered via AMI provided a channel is attached. Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.
This change also introduces the multi object blob mechanism used to send
multiple snapshot types in a single message. The dialplan app UserEvent
was also changed to use multi object blob, and a new stasis message type
created to handle them.
ASTERISK-22697 #close
Review: https://reviewboard.asterisk.org/r/3494/
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This changes fixes a crash that occurs when stasis determines if it
should send a message out to an application or not. The code
incorrectly assumed that a bridge snapshot would always be present
when in reality for failure cases it may not be.
ASTERISK-23573 #close
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