This patch addresses the following memory/ref counting leaks:
* main/devicestate.c - unsubscribe and join our devicestate message
subscription
* main/cel.c - clean up the datastore and config objects on exist
* main/parking.c - cleanup memory leak of retriever snapshot on message
payload destruction
* res/parking/parking_bridge.c - cleanup memory leak of retrieve snapshot
on message payload destruction
* main/presencestate.c - unsubscribe and join the caching topic on exit
* manager.c - properly unregister the manager action "BlindTransfer"
* sorcery.c - shutdown the threadpool on exit and dispose of any wizards
(issue ASTERISK-21906)
Reported by: John Hardin
patches:
cel.patch uploaded by jhardin (license #6512)
devicestate.patch uploaded by jhardin (license #6512)
manager.patch uploaded by jardin (license #6512)
presencestate.patch uploaded by jhardin (license #6512)
retriever-channel-snapshot.patch uploaded by jhardin (license #6512)
sorcery.patch uploaded by jhardin (license #6512)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The menuselect parser is very simple. It looks for AST_MODULE_INFO and
uses any quoted string on that line as the module summary display.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds support for stasis/sounds and stasis/sounds/{ID} queries via
the Asterisk RESTful Interface (ARI, formerly Stasis-HTTP).
The following changes have been made to accomplish this:
* A modular indexer was created for local media.
* A new function to get an ast_format associated with a file extension
was added.
* Modifications were made to the built-in HTTP server so that URI
decoding could be deferred to the URI handler when necessary.
* The Stasis-HTTP sounds JSON documentation was modified to handle
cases where multiple languages are installed in different formats.
* Register and Unregister events for formats were added to the system
topic.
(closes issue ASTERISK-21584)
(closes issue ASTERISK-21585)
Review: https://reviewboard.asterisk.org/r/2507/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch properly packs the parameters into the send fax message so that it
actually work.
Missing a ',' between two string fields can be difficult to debug, particularly
when the actual packing succeeds. Interestingly enough, this didn't actually
crash until the JSON blob we deref'd and disposed of. Since that happened in
a different thread, it was pretty tough to track down.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
By the time something extracts the pointers from ast_json_pack, the channels
will already be disposed of. This patch properly pulls the information out of
the variables and packs them into the JSON blob.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sorcery specific object information is now opaque and allocated with the object.
This means that modules do not need to be recompiled if the sorcery specific part
is changed. It also means that sorcery can store additional information on objects
and ensure it is freed or the reference count decreased when the object goes away.
To facilitate the above a generic sorcery allocator function has been added which
also ensures that allocated objects do not have a lock.
Extended fields have been added thanks to all of the above which allows specific fields
to be marked as extended, and thus simply stored as-is within the object. Type safety
is *NOT* enforced on these fields. A consumer of them has to query and ultimately perform
their own safety check. What does this mean? Extra modules can extend already defined
structures without having to modify them.
Tests have also been included to verify extended field functionality.
Review: https://reviewboard.asterisk.org/r/2585/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support
Thanks everyone!
Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two bugs.
(1) It unlocks the channel in the framehook handlers before attempting to grab
the peer from the bridge. The locking order for the bridging framework is
bridge first, then channel - having the channel locked while attempting to
obtain the bridge lock causes a locking inversion and a deadlock. This
patch bumps the channel ref count prior to releasing the lock in the
framehook to avoid lifetime issues.
Note that this does expose a subtle problem in framehooks; that is,
something could modify the framehook list while we are executing, causing
issues in the framehook list traversal that the callback executes in.
Fixing this is a much larger problem that is beyond the scope of this
patch - (a) we already unlock the channel in this particular framehook
and we haven't run into a problem yet (as modifying the framehook list
when a channel is about to perform a fax gateway would be a very odd
operation) and (b) migrating to an ao2 container of framehooks would be
more invasive at this point. See the referenced ASTERISK issue for more
information.
(2) Directly packing channel variables into a JSON object turned out to be
unsafe. A condition existed where the strings in the JSON blob were no
longer safe to be accessed if the channel object itself was disposed of.
(issue ASTERISK-21951)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This can happen when a REGISTER request is removing a contact.
(closes issue ASTERISK-21911)
Reported by: mdavenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This pulls bridge-related CEL event triggers out of the code in which
they were residing and pulls them into cel.c where they are now
triggered by changes in bridge snapshots. To get access to the
Stasis-Core parking topic in cel.c, the Stasis-Core portions of parking
init have been pulled into core Asterisk init.
This also adds a new CEL event (AST_CEL_BRIDGE_TO_CONF) that indicates
a two-party bridge has transitioned to a multi-party conference. The
reverse cannot occur in CEL terms even though it may occur in actuality
and two party bridges which receive a AST_CEL_BRIDGE_TO_CONF will be
treated as multi-party conferences for the duration of the bridge.
Review: https://reviewboard.asterisk.org/r/2563/
(closes issue ASTERISK-21564)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The WebSocket code would allocate, on the stack, a string large enough
to hold a key provided by the client, and the WEBSOCKET_GUID. If the key
is NULL, this causes a segfault. If the key is too large, it could
overflow the stack.
This patch checks the key for NULL and checks the length of the key to
avoid stack smashing nastiness.
(closes issue ASTERISK-21825)
Reported by: Alfred Farrugia
Tested by: Alfred Farrugia, David M. Lee
Patches:
issueA21825_check_if_key_is_sent.patch uploaded by Walter Doekes (license 5674)
........
Merged revisions 391560 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes the AGI AsyncAGI event put provided AGI command arguments in
the event's environment.
(closes issue ASTERISK-21304)
Patch-By: Dirk Wendland
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391271 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
The code was still attempting to pack an additional item into the blobs
that didn't exist. Crashes ensued. This patch modifies the publishing of
these messages so that the correct number of items are packed in the JSON.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ast_channel_cached_blob_create was merged,
ast_channel_blob_create_from_cache was partially removed in an
unresolved merge conflict. This restores ast_channel_blob_create_from_cache
and refactors usage of ast_channel_cached_blob_create (requires an
ast_channel) to use ast_channel_blob_create_from_cache (requires a
channel uniqueid) instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r389799, a number of fax errors in gateway mode were fixed by using the
appropriate function to get a channel's peer while in a bridge. This patch
does two things:
(1) It uses the same function in res_fax_spandsp while starting the fax
gateway. Without this, the fax gateway will not actually start up, as
res_fax_spandsp also must inspect the channel's peer in a two-party
bridge
(2) It refactors some ao2 objects in sendfax_exec to use RAII_VAR. This was
reverted in r389799 as some off nominal paths were getting hit without
the fix in (1) that indicated an ao2 object issue; this turned out to
be a red herring (which is an odd phrase)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fax gateway requires knowledge of a channel's peer in a bridge. This patch
now uses the supported mechanisms to get this information.
This is acceptable for a few reasons:
* Fax gateway can only ever work in a 2-party bridge
* Fax gateway cannot work when not in a bridge
* Fax gateway cannot work without knowledge of the capabilities of both
channels in the fax operation (it is, after all, a gateway)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Initialize a Stasis-Core message type prior to initializing a caching topic.
The caching topic will attempt to use the message type.
* Don't attempt to publish Stasis-Core messages from remote console connections.
They aren't the main process; they shouldn't attempt to behave as it (they also
don't have the infrastructure to do so)
* Don't treat a JSON object as an ao2 object (whoops)
* In asterisk.c, ref bump the JSON even package that is distributed with the
event meta data. The callers assume that they own the reference, and the packing
routine steals references.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
* ChanSpyStart/Stop
* MonitorStart/Stop
* MusicOnHoldStart/Stop
* FullyBooted/Reload
* All Voicemail/MWI related events
In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.
Review: https://reviewboard.asterisk.org/r/2532
(closes issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Both of them are covered in the dynamic parking review on
https://reviewboard.asterisk.org/r/2550 - Remove unref against
parking lot that the bridge did on dissolve since the reference
wasn't taken in the first place. On a swap, reapply bridge roles
in order to get music on hold and such playing on the channel that
swaps into the bridge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change ensures that the INVITE session remains valid for the lifetime
of the session object itself by increasing the session count on the dialog that
the INVITE session is allocated from. Once this reaches zero (normally as a result
of decrementing it within the session destructor) the dialog, and INVITE session,
are destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389609 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
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either before
it was initialized or after it had been cleaned up. It turns out that
this assertion fires during shutdown.
This actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message types
used by the subscription could be freed before the final message of
the subscription was processed.
This patch adds stasis_subscription_join(), which blocks until the
last message has been processed by the subscription. Since joining was
most commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also added.
Similar functions were also added to the stasis_caching_topic and
stasis_message_router, since they wrap subscriptions and have similar
problems.
Other code in trunk was refactored to join() where appropriate, or at
least verify that the subscription was complete before being
destroyed.
Review: https://reviewboard.asterisk.org/r/2540
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 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
This moves the JSON event generators out of the Stasis-HTTP modules and
into standalone JSON-related counterparts so that Stasis-HTTP and
res_stasis can depend on them without creating dependency cycles. This
also provides a future location for Swagger Model validator functions
once the generators for that code are written.
Review: https://reviewboard.asterisk.org/r/2534/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After the merge of support for the realtime sorcery module, extensions that
contained a pattern were not being found through odbc realtime. It was tracked
down to this one line that was advancing to the next variable list before it
should have been. The removal of this one line fixes this.
Tested this fix on my machine.
Received confirmation that this is the right fix from file on IRC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I've noticed when doing a graceful shutdown that the res_stasis_http.so
module gets unloaded before the modules that use it, which causes some
asserts during their unload.
While r386928 was a quick hack to get it to not assert and die, this
patch increases the use counts on res_stasis.so and res_stasis_http.so
properly. It's a bigger change than I expected, hence the review instead
of just committing it.
Review: https://reviewboard.asterisk.org/r/2489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a framework in res_stasis for handling events from
channel topics. JSON event generation and validation code is created
from event documentation in rest-api/api-docs/events.json to assist in
JSON event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications.
The userevent application has been refactored along with the code that
handles userevent channel blob events to pass the headers as key/value
pairs in the JSON blob. As a side-effect, app_userevent now handles
duplicate keys by overwriting the previous value.
Review: https://reviewboard.asterisk.org/r/2428/
(closes issue ASTERISK-21180)
Patch-By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we send out a CN packet (for instance, in the case of using rtpkeepalives),
we are not setting the payload code properly. Also, we are setting the marker
bit when we shouldn't be according to RFC 3389, section 4.
AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using
ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup().
11 and trunk already use the appropriate function.
* In 1.8, use ast_rtp_codecs_payload_code()
* Remove the setting of the marker bit
* Fix the debug message by incrementing the seqno after the debug message is set
in order to display the correct seqno that was sent out
(closes issue ASTERISK-21246)
Reported by: Peter Katzmann
Tested by: Peter Katzmann, Michael L. Young
Patches:
asterisk-21246-rtp-cng-payload-error_1.8_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2500/
........
Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 388112 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we first introduced the channel blob types, the JSON blobs were
self identifying by a required "type" field in the JSON object
itself. This, as it turns out, was a bad idea.
When we introduced the message router, it was useless for routing based
on the JSON type. And messages had two type fields to check: the
stasis_message_type() of the message itself, plus the type field in the
JSON blob (but only if it was a blob message).
This patch corrects that mistake by removing the required type field
from JSON blobs, and introducing first class stasis_message_type objects
for the actual message type.
Since we now will have a proliferation of message types, I introduced a
few macros to help reduce the amount of boilerplate necessary to set
them up.
Review: https://reviewboard.asterisk.org/r/2509
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An endpoint is an external device/system that may offer/accept
channels to/from Asterisk. While this is a very useful concept for end
users, it is surprisingly not a core concept within Asterisk itself.
This patch defines ast_endpoint as a separate object, which channel
drivers may use to expose their concept of an endpoint. As the channel
driver creates channels, it can use ast_endpoint_add_channel() to
associate channels to the endpoint. This updated the endpoint
appropriately, and forwards all of the channel's events to the
endpoint's topic.
In order to avoid excessive locking on the endpoint object itself, the
mutable state is not accessible via getters. Instead, you can create a
snapshot using ast_endpoint_snapshot_create() to get a consistent
snapshot of the internal state.
This patch also includes a set of topics and messages associated with
endpoints, and implementations of the endpoint-related RESTful
API. chan_sip was updated to create endpoints with SIP peers, but the
state of the endpoints is not updated with the state of the peer.
Along for the ride in this patch is a Stasis test API. This is a
stasis_message_sink object, which can be subscribed to a Stasis
topic. It has functions for blocking while waiting for conditions in
the message sink to be fulfilled.
(closes issue ASTERISK-21421)
Review: https://reviewboard.asterisk.org/r/2492/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds the ability for modules to add themselves as observers
to sorcery object types. Observers can be notified when objects are
created, updated, or deleted as well as when the object type is loaded or
reloaded. Observer notifications are done using a thread pool in a serialized
fashion so the caller of the sorcery API calls is minimally impacted.
This also adds the ability to create JSON changesets of a sorcery object.
Tests are also present to confirm all of the above functionality.
Review: https://reviewboard.asterisk.org/r/2477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In certain situations, when the RTP engine goes to send a DTMF end digit
it may be in a situation where the remote address is no longer available,
or the digit that was supposed to be sent is invalid. In such cases, we
need to clear the RTP counters appropriately. Otherwise, when the RTP
source is set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party (signficantly).
(closes issue ASTERISK-21522)
Reported by: Corey Farrell
patches:
rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909)
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Merged revisions 387213 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 387216 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed crash when res_stasis_http is unloaded before the
implementation modules.
* Cleaned up test initialization for test_stasis_http.so.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change does the following:
1. Adds the sorcery realtime module
2. Adds unit tests for the sorcery realtime module
3. Changes the realtime core to use an ast_variable list instead of variadic arguments
4. Changes all realtime drivers to accept an ast_variable list
Review: https://reviewboard.asterisk.org/r/2424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the various res_sip modules with their proper menuselect
options and proper dependencies, such that Asterisk still has a snowball's
chance in hell of compiling without pjproject.
Much thanks to snuffy(-home|-work) for making everyone's life
easier with this patch.
Review: https://reviewboard.asterisk.org/r/2472/
(closes issue ASTERISK-21669)
Reported by: snuffy
patches:
xml-depends.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to get people familiar with the Stasis message bus, it would
be useful to have something of a tutorial. Since I'm not clever enough
to think of some cool integration we could do with Twitter, I settled
for something that might actually be useful.
This patch adds a res_statsd.so module, which implements a basic
statsd[1] client. Statsd is a very simple statistics gathering server,
which can publish its results to a backend graphing engine, like
Graphite[2]. There are several different Statsd server
implementations[3], so you can pick what works best for your
environment.
The actual example of how to use the Stasis message bus is in
res_chan_stats.so. This module demonstrates how to use subscriptions
and the message router by monitoring messages and posting channels
stats to the statsd server.
A wiki page walking through res_chan_stats.so is forthcoming.
[1]: https://github.com/etsy/statsd/
[2]: http://graphite.readthedocs.org/en/latest/
[3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/
Review: https://reviewboard.asterisk.org/r/2460/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.
SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.
API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.
The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates. The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.
The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.
(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were several reports of deadlock when using
res_timing_pthread. Backtraces indicated that one thread was blocked
waiting for the write to the pipe to complete and this thread held
the container lock for the timers. Therefore any thread that wanted
to create a new timer or read an existing timer would block waiting
for either the timer lock or the container lock and deadlock ensued.
This patch changes the way the pipe is used to eliminate this source
of deadlocks:
1) The pipe is placed in non-blocking mode so that it would never
block even if the following changes someone fail...
2) Instead of writing bytes into the pipe for each "tick" that's
fired the pipe now has two states--signaled and unsignaled. If
signaled, the pipe is hot and any pollers of the read side
filedescriptor will be woken up. If unsigned the pipe is idle. This
eliminates even the chance of filling up the pipe and reduces the
potential overhead of calling unnecessary writes.
3) Since we're tracking the signaled / unsignaled state, we can
eliminate the exta poll system call for every firing because we know
that there is data to be read.
(closes issue ASTERISK-21389)
Reported by: Matt Jordan
Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis
patches:
0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417)
(closes issue ASTERISK-19754)
Reported by: Nikola Ciprich
(closes issue ASTERISK-20577)
Reported by: Kien Kennedy
(closes issue ASTERISK-17436)
Reported by: Henry Fernandes
(closes issue ASTERISK-17467)
Reported by: isrl
(closes issue ASTERISK-17458)
Reported by: isrl
Review: https://reviewboard.asterisk.org/r/2441/
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This patch adds the concept of ast_websocket_server to
res_http_websocket, allowing WebSocket connections on URL's more more
than /ws.
The existing funcitons for managing the WebSocket subprotocols on /ws
still work, so this patch should be completely backward compatible.
(closes issue ASTERISK-21279)
Review: https://reviewboard.asterisk.org/r/2453/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_enable_distributed_devstate is no longer applicable to how the
distributed device state system works and is no longer necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385886 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the move from Asterisk's event system to Stasis, this makes
distributed device state aggregation always-on, removes unnecessary
task processors where possible, and collapses aggregate and
non-aggregate states into a single cache for ease of retrieval. This
also removes an intermediary step in device state aggregation.
Review: https://reviewboard.asterisk.org/r/2389/
(closes issue ASTERISK-21101)
Patch-by: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After some discussion on asterisk-dev, it was decided that the bulk of
the logic in app_stasis actually belongs in a resource module instead
of the application module.
This patch does that, leaves the app specific stuff in app_stasis, and
fixes up everything else to be consistent with that change.
* Renamed test_app_stasis to test_res_stasis
* Renamed app_stasis.h to stasis_app.h
* This is still stasis application support, even though it's no
longer in an app_ module. The name should never have been tied to
the type of module, anyways.
* Now that json isn't a resource module anymore, moved the
ast_channel_snapshot_to_json function to main/stasis_channels.c,
where it makes more sense.
Review: https://reviewboard.asterisk.org/r/2430/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.
(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
rtp-timestamp.patch uploaded by pbertera (License 5943)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.
This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.
Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.
(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch sets the protocols container provided by res_http_websocket to NULL
when the module gets unloaded and adds the necessary checks when adding/
removing a websocket protocol. This prevents some FRACKing on an invalid
pointer to the disposed container if a module that uses res_http_websocket is
unloaded after it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.
This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.
Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.
Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.
Other changes along for the ride are:
* An ast_frame_dtor function that's RAII_VAR safe
* Some common JSON encoders for name/number, timeval, and
context/extension/priority
Review: https://reviewboard.asterisk.org/r/2361/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Updated test_uuid.c to test the new API call.
* Made system use the new API call to eliminate "10's of lines" where
used.
* Fixed untested ast_strdup() return in stasis_subscribe() by eliminating
the need for it. struct stasis_subscription now contains the uniqueid[]
string.
* Fixed some issues in exchangecal_write_event():
Create uid with enough space for a UUID string to avoid a realloc.
Fix off by one error if the calendar event provided a UUID string.
There is no need to check for NULL before calling ast_free().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.
(closes issue ASTERISK-21323)
Reported by: andrea
Tested by: Kinsey Moore, andrea, John Bigelow
Patches:
whitenoise_fix.diff uploaded by Kinsey Moore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The format attribute resource for H.264 video performs an unsafe read against a
media attribute when parsing the SDP. The value passed in with the format
attribute is not checked for its length when parsed into a fixed length buffer.
This patch resolves the vulnerability by only reading as many characters from
the SDP value as will fit into the buffer.
(closes issue ASTERISK-20901)
Reported by: Ulf Harnhammar
patches:
h264_overflow_security_patch.diff uploaded by jrose (License 6182)
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This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.
Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matthew Jordan
David M. Lee
Richard Mudgett
Kinsey Moore
Joshua Colp
Jonathan Rose
Jason Parker
Mark Michelson
Michael L. Young
Alec L Davis
Sean Bright