The original issue noted that the bridge is orphaned when res_parking.so
is not loaded and a call uses the dial kK flags.
A similar issue happens when only one of the park flags is used. In this
case you have the bridge with one or the other channel left in it. The
channel and bridge will stay around until the channel hangs up.
* Fixed the initial bridge channel push failure to act as if the channel
were kicked out of the bridge. The bridge then decides if it needs to be
dissolved.
(closes issue ASTERISK-22629)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2928/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge. With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.
* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.
* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper(). The call to
ast_translator_best_choice() got them backwards.
* Updated some callers of ast_channel_make_compatible() and the function
documentation. There is actually a difference between the two channels
passed in.
* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible(). The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.
(closes issue ASTERISK-22542)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2915/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.
* GET /applications - list current applications
* GET /applications/{applicationName} - get details of a specific application
* POST /applications/{applicationName}/subscription - explicitly subscribe to
a channel, bridge or endpoint
* DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
from a channel, bridge or endpoint
Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.
Review: https://reviewboard.asterisk.org/r/2862
(issue ASTERISK-22451)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a party leaves a bridge, there may be more participants in the bridge than expected.
As such, it is important not to make assumptions regarding the list of channels in a
bridge.
This change makes it so that when a party leaves a native RTP bridge, we unbridge it and
the party it was bridged with. Previously, the first and last channels in the list were
unbridged since it was assumed that these were the two channels that had been bridged. As
previously stated, a new party had been inserted into the bridge, so this logic did not
work properly.
(closes issue ASTERISK-22615)
reported by Matt Jordan
(closes issue ASTERISK-22532)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2899
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This introduces usage of an additional libxslt cleanup function,
xsltCleanupGlobals, when the configure script detects that it is
available. Early versions of the library did not include this function.
(closes issue ASTERISK-22570)
Reported by: Corey Farrell
Patches:
xsltCleanupGlobals.patch uploaded by Corey Farrell (License 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since caches are updated on publisher threads, there is no need
to wait for the cache updates to occur after a stasis message
is published.
In the case of chan_pjsip device state changes, this set of
changes caused an improvement to performance.
Review: https://reviewboard.asterisk.org/r/2890
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change introduces the ability to stage channel snapshot
creation and publishing by suppressing the implicit creation
and publishing that some functions have. Once all operations
are executed the staging is marked as done and a single snapshot
is created and published.
Review: https://reviewboard.asterisk.org/r/2889/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback. During state change it was assumed
what was in the mod_data was nothing or the response callback. However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.
Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.
(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
OBJ_CONTINUE was a strange feature that came into the world under
suspicious circumstances to support an abuse of the ao2_container by
chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is safe to
remove it.
The simplified code should help performance slightly and make
understanding the code easier.
Review: https://reviewboard.asterisk.org/r/2887/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes the CEL peer field useful again for BRIDGE_ENTER and
BRIDGE_EXIT events and fills the field with a comma-separated list of
all channels in the bridge other than the channel that is entering or
exiting the bridge.
Review: https://reviewboard.asterisk.org/r/2840/
(closes issue ASTERISK-22393)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While handling a registration request a race condition could occur if/when two+
clients registered at the same time. This happened when one request obtained a
copy of the current contacts for an AOR and another request did the same before
the first request updated. Thus the second would update and overwrite the first
(or vice-versa depending on which actually updated first). In the case of it
being the same contact two "add" events would be raised.
pjsip registration handling is now serialized to alleviate this issue.
(closes issue AST-1213)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2860/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The endpoint option does not apply to communication with external entities. Rather,
the option is applied to all communications with the endpoint. The external_media_address
transport configuration option may override the endpoint option if it turns out that
we are going to be communicating with an external entity.
Two things of note:
1) I have not updated the XML documentation. This is being taken care of by Rusty as part
of his work on issue ASTERISK-22405
2) This commit is likely to cause testsuite failures since there are tests that use the
external_media_address endpoint option, and they will need to be changed over. Well, I'm
planning to get that updated ASAP after this commit.
(closes issue ASTERISK-22528)
reported by Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This was an unintentional consequence of the update of features.conf to use the
config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk
developers list for pointing out the problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.
With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).
This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).
(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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Merged revisions 398649 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed resulting warnings with improper use of ao2_global_obj_replace().
* Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the
equivalent and more appropriate ao2_global_obj_release() call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some configuration objects currently won't place nice if reloaded.
Specifically, in this case the pjsip transport objects. Now when
registering an object in sorcery one may specify that the object is
allowed to be reloaded or not. If the object is set to not reload
then upon reloading of the configuration the objects of that type
will not be reloaded. The initially loaded objects of that type
however will remain.
While the transport objects will not longer be reloaded it is still
possible for a user to configure an endpoint to an invalid transport.
A couple of log messages were added to help diagnose this problem if
it occurs.
(closes issue ASTERISK-22382)
Reported by: Rusty Newton
(closes issue ASTERISK-22384)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2807/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.
The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.
There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.
The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)
(closes issue ASTERISK-21582)
Review: https://reviewboard.asterisk.org/r/2693/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The config options test requires the entire configuration item to be transparent from
the documentation system. So we let it do that too.
As an aside, please do not use this power for evil. Documentation is your friend, and
you really should document your configurations. Hiding your module's configuration
information from the system attempting to enforce some sanity in the universe is something
only a Bond villain would contemplate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.
The following cases need to be handled when a channel is moved around in
the system.
* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.
* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)
* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.
* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.
The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.
(closes issue ASTERISK-22043)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2791/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.
* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().
* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.
* Fixed some formatting in ast_bt_get_symbols().
* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.
* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.
* Moved __dump_backtrace() because of compile issues with the utils
directory.
(closes issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2778/
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Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds error checking to ARI bridge operations, when
adding/removing channels to/from bridges.
In general, the error codes fall out as follows:
* Bridge not found - 404 Not Found
* Bridge not in Stasis - 409 Conflict
* Channel not found - 400 Bad Request
* Channel not in Stasis - 422 Unprocessable Entity
* Channel not in this bridge (on remove) - 422 Unprocessable Entity
(closes issue ASTERISK-22036)
Review: https://reviewboard.asterisk.org/r/2769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are times when a configuration option should not have documentation.
1. Some options are registered with a particular object merely as a warning to
users. These options aren't even really 'deprecated' - which has its own
separate API call - they are actually provided by a different configuration
file. The options are merely registered so that the user gets a warning that
a different configuration file provides the item.
2. Some object types - most notably some used by modules that use sorcery - are
completely internal and should never be shown to the user.
3. Sorcery itself has several 'hidden' fields that should never be shown to a
user.
This patch updates the configuration framework and sorcery with additional API
calls that allow a module to register types as internal and options as not
requiring documentation. This bypasses the XML documentation checking.
This patch also re-enables the strict XML documentation checking in trunk, as
well as updates some documentation that was missing.
Review: https://reviewboard.asterisk.org/r/2785/
(closes issue ASTERISK-22359)
Reported by: Matt Jordan
(closes issue ASTERISK-22112)
Reported by: Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds ARI functions to be able to turn on/off music on hold in a
bridge. It actually functions more as a background music without
further actions on the bridge since if the rest of the channels
in the bridge aren't explicitly muted, they will still be able
to communicate.
(closes issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2688/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.
* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.
(closes issue ASTERISK-22042)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2772/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This essentially makes app_queue usable again. From reviewboard:
* Reporting of transfers and call completion is done by creating stasis
subscriptions and listening for specific events in order to determine
when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
Mixmonitor API now instead of using ast_pbx_run()
In addition to the changes in app_queue, there are several supplementary changes as well:
* Queue logging now differentiates between attended and blind transfers. A
note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
includes which of the two local channels involved is the destination of
the optimization, the channel that is replacing the destination local channel,
and an identifier so that begin and end events can be matched to each other.
The end events are now sent whether the optimization was successful or not and
includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
be set on a bridge. This is necessary because the queue requires that its
bridge only allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517)
Reported by Matt Jordan
(closes issue ASTERISK-21943)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2694
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
