This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some regex implementations won't compile an empty string. Assuming that
it's equivalent of a regex that will match anything, use ".?" instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds support for externally initiated parking requests. In particular,
chan_skinny has a protocol level message that initiates a call park.
This patch now supports that option, as well as the protocol specific
mechanisms in chan_dahdi/sig_analog and chan_mgcp.
* A parking bridge features virtual table has been added that provides
access to the parking functionality that the Bridging API needs. This
includes requests to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to a parking
extension, determine if an extension is a parking extension, as well as the
actual "do the parking" request from the Bridging API.
* Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
functions
* The removal of some - but not all - dead parking code from features.c
This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)
Review: https://reviewboard.asterisk.org/r/2710
(closes issue ASTERISK-22134)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Subversion doesn't do quote processing, so it actually thinks that the
closing quote in 'Revision"' is a part of the keyword.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a module reload of core components causes specific functions
compiled into the Asterisk binary to be reloaded. The table of said functions
was still pointing to the old features reload mechanism, and not the new one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This renames all files and API calls from several variants of
Stasis-HTTP to ARI including:
* Stasis-HTTP -> ARI
* STASIS_HTTP -> ARI
* stasis_http -> ari (ast_ari for global symbols, file names as well)
* stasis http -> ARI
Review: https://reviewboard.asterisk.org/r/2706/
(closes issue ASTERISK-22136)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most hook callbacks did not need the bridge parameter. The pointer value
could become invalid if the channel is moved to another bridge while it is
executing.
* Fixed some issues in feature_attended_transfer() as a result.
* Reduce the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the bridge
channel hooks.
* Removed basic bridge requirement on feature_blind_transfer(). It does
not require the basic bridge like feature_attended_transfer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed feature limits to not use special members of struct
ast_bridge_features.
* Fixed memory leak in off nominal paths of bridge_builtin_set_limits().
* Fixed off nominal path in ast_bridge_features_limits_construct() freeing
unallocated memory if it was not called by bridge_builtin_set_limits().
* Made bridge_builtin_interval_features.so unloadable.
* Simplified parking's use of its duration interval hook.
* Made BridgeWait S option not depend upon another module being loaded.
(closes issue ASTERISK-22107)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2701/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes arguments for BridgeWait from BridgeWait(role, options) to
BridgeWait(bridge_name, role, options). Now multiple holding bridges may
be created and referenced by this application.
(closes issue ASTERISK-21922)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2642/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes the previously #if 0'd code. The functionality removed has either
been subsumed by the Bridging API or is no longer applicable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Convert interval timers to use the ast_waitfor_nandfds() timeout.
* Remove bridge channel action for intervals. Now the main loop handles
running interval hooks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Reduced the number of hook containers to just dtmf_hooks,
interval_hooks, and other_hooks. As a result, several functions dealing
with the different hook containers could be combined.
* Extended the generic hook struct for DTMF and interval hooks instead of
using a variant record.
* Merged the special talk detector hook into the other_hooks container.
* Replaced ast_bridge_features_set_talk_detector() with
ast_bridge_talk_detector_hook().
(issue ASTERISK-22107)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add an error message so you know when a feature is not available and you
tried to use it. It usually means the module has not been loaded.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ensure that the BridgeInfo command provides adequate state information
about channels by publishing the full channel snapshot for
BridgeInfoChannel subevents. This prevents a two-stage lookup since
most consumers will be keying on channel names instead of uniqueids.
(closes issue ASTERISK-22140)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It pulls out bridge_channel and puts it into its own translation unit
* It adds public and protected headers for bridging_channel. Protected
functions are appropriate only for the Bridging API and sub-classes of a
bridge.
(issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created a native_dahdi bridging technology for use with the new bridging
API.
The new bridging technology is part of the chan_dahdi channel driver
because it is very specific to that driver. Rather than include the new
code directly into chan_dahdi.c the new bridge technology is in its own
file and linked into chan_dahdi.so. A large part of this change is the
mechanical process of moving declarations around so chan_dahdi.c can be
split up into more files later.
* Changed the bridging core to pass NULL frames into the channel
technologies instead of discarding them. The channel technologies may
need the proding to determine if their configuration is still valid.
(closes issue ASTERISK-21886)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This greatly modifies the operation of DTMF attended transfers so that
the full range of options from features.conf applies.
In addition, a new option has been added that allows for a transferer
to switch between bridges during a transfer before completing the
transfer.
(closes issue ASTERISK-21543)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2654
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In previous versions of Asterisk, the zombies roamed freely,
unchecked and uncontrolled. They ravaged Asterisk systems with
their biting and their nashing and their pointy teeth.
Sometimes, you couldn't even hang them up.
Now, zombies are rare. They still *technically* exist in certain
places, but they are controlled. Kind of like a zombie zoo: you can
see them, but you can't touch them, and they can't touch you.
Bring your kids!
Because zombies are now population controlled with a very short lifespan,
there's no reason to rename the channels to '%s<ZOMBIE>'. The channels
are guaranteed to die off quickly; the rename really is just confusing
at this point.
This patch finally removes the renaming. On the plus side: this made
my life easier in CDRs during call pickup and attended transfers to
an Asterisk application. It will make other folks lives easier as well!
Review: https://reviewboard.astierks.org/r/2690/
(closes issue ASTERISK-21699)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The stasis_cache_update messages are somewhat cumbersome to handle
with the stasis_message_router. Since all updates have the same
message type, they are normally handled with the same route.
Since caching itself is a first class component of stasis-core, it
makes sense for the router to handle the cache update messages itself.
This patch adds stasis_message_router_add_cache_update() and
stasis_message_router_remove_cache_update() to handle the routing of
stasis_cache_update messages.
This patch also corrects an issue with manager_{bridging,channels}.c,
where events might be reordered. The reordering occurs because the
components use different message routers, which they needed because
they both needed to route cache update messages. They now both use
manager's router, and add cache routes for just the cache updates they
are interested in.
(closes issue ASTERISK-22038)
Review: https://reviewboard.asterisk.org/r/2677/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies parsing of cookies in Asterisk's http server by doing an
explicit comparison of the "Cookie" header instead of looking at the first
6 characters to determine if the header is a cookie header. This avoids
parsing "Cookie2" headers and overwriting the previously parsed "Cookie"
header.
Note that we probably should be appending the cookies in each "Cookie"
header to the parsed results; however, while clients can send multiple
cookie headers they never really do. While this patch doesn't improve
Asterisk's behavior in that regard, it shouldn't make it any worse either.
Note that the solution in this patch was pointed out on the issue by the
issue reporter, Stuart Henderson.
(closes issue ASTERISK-21789)
Reported by: Stuart Henderson
Tested by: mjordan, Stuart Henderson
........
Merged revisions 394899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 394900 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies manager to allow the allowmultiplelogin setting to be set
on an account by account basis. When set in the general context, it will act
as the default for the defined accounts. Setting it in the account will
override the general setting.
(closes issue ASTERISK-21324)
Reported by: vldmr
patches:
asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent
local channel optimizations. Local channel optimizations were one of
several things conveyed by the now defunct BRIDGE_UPDATE event type.
This also adds a unit test to test generation of this new CEL event.
Review: https://reviewboard.asterisk.org/r/2676/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty()
NULL tolerant.
* Made ast_audiohook_detach_list() return void since it is a destructor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds new flags to the channel tech properties that flag it as
different types of implementation detail used exclusively to provide a
feature. Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are the
only two marked as such by this patch.
Review: https://reviewboard.asterisk.org/r/2633/
(closes issue ASTERISK-21873)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)
(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols being
compared after the string length calculation such that it shows the correct
string length in the output
2. It adds some additional debug that displays when it matches on a
sub-protocol and when it fails
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing asserts
in the /stasis/core tests.
If the caching topic's subscription is the last reference to the
caching topic, it will destroy the caching topic after the final
message has been processed. When dispatching to a different thread,
this usually gave the unsubscribe enough time to finish before
destruction happened. Now, however, it consistently destroys before
unsubscription is complete.
This patch adds an extra reference to the caching topic, to hold it
for the duration of the unsubscription.
This patch also removes an extra unref that was happening when the
final message was received by the caching topic. It was put there
because of an extra ref that was put into the caching topic's
constructor. Both have been removed, which makes the destructor a bit
less confusing.
Review: https://reviewboard.asterisk.org/r/2675/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel is hungup, both an APP_END event and a HANGUP event can be
fired. To ensure that HANGUP events occur after APP_END events, the method
callbacks for the APP_END event should be processed prior to the callbacks
for the HANGUP event.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch attempts to fix some possible race conditions in shutdown of the
CDR engine. It:
* Adds a cleanup handler to only unsubscribe and join on stasis messages during
graceful shutdown. The cleanup handler should execute before the regular atexit
handler, as we want to unsubscribe for any further messages before dispatching
the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Peter J Philipp pointed out that there are two checks that ensure that len is
not less than 0. If len is less than 0, the function returns. Having both of
them is clearly redundant.
This removes the second and attempts to clarify (slightly) the error condition.
(closes issue ASTERISK-21772)
Reported by: Peter J Philipp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It simplifies the Dial handling in CDRs. As a rule, the caller in a dial
relationship is always the Party A. There was some logic present in the
handling of the dial message that could, conceivably, pick the caller
as Party A for the beginning of the dial and the peer as Party A for the
end of the dial. This shouldn't have happened if the code in the bridging
framework was doing its job; however, that was broken and it led to the
FRACK. As it is, this code was overly ocmplex and not needed: the caller,
if present, should always be Party A. Period.
* It properly checks to see if a channel will continue on in the dialplan.
ast_check_hangup - much like cake at the end - is a lie. It will tell
you that you are hungup when you are not. Do not believe it.
I would make this function tell the truth, but I'm nervous that we've been
depending on it sitting on its throne of lies for far too long, and it would
probably break lots of things. So I'm just checking the "internal" soft
hangup flags, like everyone else.
(closes issue ASTERISK-22060)
Reported by: Mark Michelson
(issue ASTERISK-21831)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a virtual table of callbacks to core_unreal. These callbacks can be
supplied by concrete implementations of "unreal" channel drivers, which lets
the unreal channel driver call specific functionality when it performs some
action. Currently, this is done to notify implementations when an
optimization operation has begun, and when an optimization operation has
succeeded.
* It adds Stasis-Core messages for Local channel bridging and Local channel
optimization. Local channel optimization is now two events: a Begin and an
End. Some consumers of Stasis-Core may want to know when an operation is
beginning so that they can 'prepare' their information; others will be more
concerned about when the operation has completed, so that they can 'fix up'
information. Stasis-Core allows for both, as does AMI.
Review: https://reviewboard.asterisk.org/r/2552
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch reorders certain actions that may raise Stasis messages in the
channel destructor such that they occur before the Stasis cache is cleared.
Once the Stasis cache is cleared, its rather a bad idea to be trying to
publish information about a channel.
(closes issue ASTERISK-22001)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
channel is executing dialplan hangup logic, i.e., the 'h' extension or a
hangup handler. Stasis messages now also convey the soft hangup flag so
consumers of the messages can know when a channel is executing said
hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
and other consumers of Stasis have been updated to look for this flag to
know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
leaving a bridge. Previously, a new CDR was automatically created when a
channel left a bridge and put into the 'Pending' state; however, this
way of handling CDRs made it difficult for the 'endbeforehexten' logic to
work correctly - there was always a new CDR waiting in the hangup logic
and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
hangup routine. This patch completely removes the Pending state and instead
defers creation of the new CDR until it gets a new message that requires
a new CDR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
information in the RTCP events. Because Stasis provides a cache, Jaco's
patch was modified to pass the channel uniqueid to the RTP layer as
opposed to a pointer to the channel. This has the following benefits:
(1) It keeps the RTP engine 'clean' of references back to channels
(2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast) could also
raise RTCP information. The engine provides structs to represent RTCP headers
and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
raise an event when we sent a RR report.
Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.
Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.
Review: https://reviewboard.asterisk.org/r/2603/
(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
(closes issue ASTERISK-21471)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Legacy channel drivers often include the ability to set a default parking lot
on an endpoint basis; when channels are created for that endpoint, they inherit
the parkinglot option. Parking used to use this option more frequently; while
it is still supported, other options (such as using channel variables or
creation of a custom parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly useful - it
is rarely (if ever) changed on a channel and some consumers of channel
snapshots, such as ARI, will never use the information.
(closes issue ASTERISK-21968)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.
(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes a few minor bugs and one major one: the CDR by bridge
container was less than helpful. The mechanism previously used to try
and find all of the CDRs in a particular bridge ended up missing CDRs,
resulting in incorrect records.
When looking up CDRs in a bridge, we now just bite the bullet and do
a selection across all existing CDRs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While a Stasis configuration file is nice, it shouldn't be mandatory.
We can carry on with default values.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, the order of procedures on a bridge push was
* Add new bridge channel to bridge's array.
* Pull the swap channel out of the bridge
* Publish a bridge enter event.
The problem is that when the swap channel was pulled from the bridge,
a bridge leave event would be published. The bridge snapshot
published during the bridge leave showed the new channel that had
been added to the bridge, but there had been no bridge enter event
for that channel.
The fix provided here was to change the order a bit
* Add new bridge channel to bridge's array.
* Publish bridge enter event.
* Pull the swap channel out of the bridge.
This makes it so that the bridge snapshots during the stasis
events are accurate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the first step in adding recording support to the
Asterisk REST Interface.
Recordings are stored in /var/spool/recording. Since recordings may be
destructive (overwriting existing files), the API rejects attempts to
escape the recording directory (avoiding issues if someone attempts to
record to ../../lib/sounds/greeting, for example).
(closes issue ASTERISK-21594)
(closes issue ASTERISK-21581)
Review: https://reviewboard.asterisk.org/r/2612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The appropriate settings for the Stasis threadpool is very system
specific, depending upon both workload and system configuration.
This patch adds a stasis.conf file which can be used to configure the
key attributes of the threadpool for the Stasis message bus.
(closes issue ASTERISK-21280)
Review: https://reviewboard.asterisk.org/r/2651/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds authentication support to ARI.
Two authentication methods are supported. The first is HTTP Basic
authentication, as specified in RFC 2617[1]. The second is by simply
passing the username and password as an ?api_key query parameter
(which allows swagger-ui[2] to authenticate more easily).
ARI usernames and passwords are configured in the ari.conf file
(formerly known as stasis_http.conf). The user may be set to
`read_only`, which will prohibit the user from issuing POST, DELETE,
etc. Also, the user's password may be specified in either plaintext,
or encrypted using the crypt() function.
Several other notes about the patch.
* A few command line commands for seeing ARI config and status were
also added.
* The configuration parsing grew big enough that I extracted it to
its own file.
[1]: http://www.ietf.org/rfc/rfc2617.txt [2]:
https://github.com/wordnik/swagger-ui
(closes issue ASTERISK-21277)
Review: https://reviewboard.asterisk.org/r/2649/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started with the simple idea of changing the /events data
model to be more sane. The original model would send out events like:
{ "stasis_start": { "args": [], "channel": { ... } } }
The event discriminator was the field name instead of being a value in
the object, due to limitations in how Swagger 1.1 could model objects.
While technically sufficient in communicating event information, it was
really difficult to deal with in terms of client side JSON handling.
This patch takes advantage of a proposed extension[1] to Swagger which
allows type variance through the use of a discriminator field. This had
a domino effect that made this a surprisingly large patch.
[1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ
In changing the models, I also had to change the swagger_model.py
processor so it can handle the type discriminator and subtyping. I took
that a big step forward, and using that information to generate an
ari_model module, which can validate a JSON object against the Swagger
model.
The REST and WebSocket generators were changed to take advantage of the
validators. If compiled with AST_DEVMODE enabled, JSON objects that
don't match their corresponding models will not be sent out. For REST
API calls, a 500 Internal Server response is sent. For WebSockets, the
invalid JSON message is replaced with an error message.
Since this took over about half of the job of the existing JSON
generators, and the .to_json virtual function on messages took over the
other half, I reluctantly removed the generators.
The validators turned up all sorts of errors and inconsistencies in our
data models, and the code. These were cleaned up, with checks in the
code generator avoid some of the consistency problems in the future.
* The model for a channel snapshot was trimmed down to match the
information sent via AMI. Many of the field being sent were not
useful in the general case.
* The model for a bridge snapshot was updated to be more consistent
with the other ARI models.
Another impact of introducing subtyping was that the swagger-codegen
documentation generator was insufficient (at least until it catches up
with Swagger 1.2). I wanted it to be easier to generate docs for the API
anyways, so I ported the wiki pages to use the Asterisk Swagger
generator. In the process, I was able to clean up many of the model
links, which would occasionally give inconsistent results on the wiki. I
also added error responses to the wiki docs, making the wiki
documentation more complete.
Finally, since Stasis-HTTP will now be named Asterisk REST Interface
(ARI), any new functions and files I created carry the ari_ prefix. I
changed a few stasis_http references to ari where it was non-intrusive
and made sense.
(closes issue ASTERISK-21885)
Review: https://reviewboard.asterisk.org/r/2639/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Refactored the AMI events in AOC onto Stasis-Core. The ast_aoc_manager_event
function now publishes a channel snapshot, along with a JSON blob describing
the advice of charge. A "to_ami" handler has also been added that converts
the channel snapshot and AOC event data back into the appropriate data structure
for use with AMI.
(closes issue ASTERISK-21472)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2643/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds several unit tests for CEL functionality and provides the
requisite framework for creating additional unit tests.
This also cleans up some reference leaks that were occurring in
Stasis-Core message callback code.
Review: https://reviewboard.asterisk.org/r/2646/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The originate APIs allow callers to provide a pointer to a channel that will
point to the originated channel if the function call succeeds. This is used by AMI
to provide channel information when the originate is performed synchronously.
Unfortunately, if the originate fails in certain ways, the outbound channel is
already disposed of during the dialing itself. This results in the channel being
improperly dereferenced by the internal originate function in pbx.c.
This patch ref bumps the channel to prevent this from occurring. Callers must now
unlock and unref the channel (which is more in line with general channel management
guidelines anyway).
This only affects manager, as it is the only consumer of this API function that
actually passes in a channel pointer.
Review: https://reviewboard.asterisk.org/r/2617/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In addition to porting those features, they now enjoy greater feature parity
with one another. Specifically, AutoMixMon now has a start and stop
message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and
TOUCH_MIXMONITOR_MESSAGE_STOP.
(closes issue ASTERISK-21553)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2620/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Originated channels are a bit odd - they are technically a dialed channel (thus
the party B or peer) but, since there is no caller, they are treated as the
party A. When entering into a bridge that already contains participants, the CDR
engine - if the CDR record is in the Dial state - attempts to match the person
entering the bridge with an existing participant. The idea is that if you dialed
someone and the person you dialed is already in the bridge, you don't need a new
CDR record, the existing CDR record describes the relationship.
Unfortunately, for an originated channel, there is no Party B. If no one was in
the bridge this didn't cause any issues; however, if participants were in the
bridge the CDR engine would attempt to match a non-existant Party B on the
channel's CDR record and explode.
This patch fixes that, and a unit test has been added to cover this case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Kinsey Moore
Matthew Jordan
David M. Lee
Mark Michelson
Richard Mudgett
Jonathan Rose
Jason Parker
Joshua Colp
Kevin Harwell