When a smart bridge operation occurs and a bridge transitions from one
technology to another the old technology is provided the channels formerly
in it and told that they are leaving. Unfortunately the bridge provided
along with them is incomplete. The bridge, despite there being channels in it,
contains none. This forces technology implementations to have additional
logic when channels are leaving or to store their own duplicated
state.
This change makes the bridge more complete so it contains the expected
channels. Now that the bridge is complete special logic within
bridge_native_rtp is no longer needed and has been removed.
Review: https://reviewboard.asterisk.org/r/4057/
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On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
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When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.
This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.
ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "bridge destroy" CLI command is invasive to bridges and can leave them in an unexpected
state for the users of them. Since this command may be useful for developers it is now
only available when developer mode is available. To take its place "all" has been added
as a valid option to the "bridge kick" CLI command. It will kick all of the channels
in the bridge out.
ASTERISK-23987
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3840/
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The previous behavior was to simply set the accountcode of an outgoing
channel to the accountcode of the channel initiating the call. It was
done this way a long time ago to allow the accountcode set on the SIP/100
channel to be propagated to a local channel so the dialplan execution on
the Local;2 channel would have the SIP/100 accountcode available.
SIP/100 -> Local;1/Local;2 -> SIP/200
Propagating the SIP/100 accountcode to the local channels is very useful.
Without any dialplan manipulation, all channels in this call would have
the same accountcode.
Using dialplan, you can set a different accountcode on the SIP/200 channel
either by setting the accountcode on the Local;2 channel or by the Dial
application's b(pre-dial), M(macro) or U(gosub) options, or by the
FollowMe application's b(pre-dial) option, or by the Queue application's
macro or gosub options. Before Asterisk v12, the altered accountcode on
SIP/200 will remain until the local channels optimize out and the
accountcode would change to the SIP/100 accountcode.
Asterisk v1.8 attempted to add peeraccount support but ultimately had to
punt on the support. The peeraccount support was rendered useless because
of how the CDR code needed to unconditionally force the caller's
accountcode onto the peer channel's accountcode. The CEL events were thus
intentionally made to always use the channel's accountcode as the
peeraccount value.
With the arrival of Asterisk v12, the situation has improved somewhat so
peeraccount support can be made to work. Using the indicated example, the
the accountcode values become as follows when the peeraccount is set on
SIP/100 before calling SIP/200:
SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200
acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200
peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100
If a channel already has an accountcode it can only change by the
following explicit user actions:
1) A channel originate method that can specify an accountcode to use.
2) The calling channel propagating its non-empty peeraccount or its
non-empty accountcode if the peeraccount was empty to the outgoing
channel's accountcode before initiating the dial. e.g., Dial and
FollowMe. The exception to this propagation method is Queue. Queue will
only propagate peeraccounts this way only if the outgoing channel does not
have an accountcode.
3) Dialplan using CHANNEL(accountcode).
4) Dialplan using CHANNEL(peeraccount) on the other end of a local
channel pair.
If a channel does not have an accountcode it can get one from the
following places:
1) The channel driver's configuration at channel creation.
2) Explicit user action as already indicated.
3) Entering a basic or stasis-mixing bridge from a peer channel's
peeraccount value.
You can specify the accountcode for an outgoing channel by setting the
CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
applications. Queue adds the wrinkle that it will not overwrite an
existing accountcode on the outgoing channel with the calling channels
values.
Accountcode and peeraccount values propagate to an outgoing channel before
dialing. Accountcodes also propagate when channels enter or leave a basic
or stasis-mixing bridge. The peeraccount value only makes sense for
mixing bridges with two channels; it is meaningless otherwise.
* Made peeraccount functional by changing accountcode propagation as
described above.
* Fixed CEL extracting the wrong ie value for the peeraccount. This was
done intentionally in Asterisk v1.8 when that version had to punt on
peeraccount.
* Fixed a few places dealing with accountcodes that were reading from
channels without the lock held.
AFS-65 #close
Review: https://reviewboard.asterisk.org/r/3601/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows the current owner of a channel to define various
feature hooks to be made available once the channel has entered a
bridge. This includes any hooks that are setup on the
ast_bridge_features struct such as DTMF hooks, bridge event hooks
(join, leave, etc.), and interval hooks.
Review: https://reviewboard.asterisk.org/r/3649/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When bridge locking was added for bridge snapshot creation, some
locations where bridge locking was added were not guaranteed to
actually have a bridge and locking NULL AO2 objects tends to cause
segfaults. This ensures that NULL bridges aren't locked.
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds the following AMI events, closely following their CLI counterparts:
BridgeDestroy
BridgeKick
BridgeTechnologyList
BridgeTechnologySuspend
BridgeTechnologyUnsuspend
BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one
channel off the bridge. When kicking a channel, specifying the bridge
also (optional) insures it is not removed from the wrong bridge. The
BridgeTechnology events allow viewing and changing suspension status,
which affects only subsequent not active bridging.
(closes ASTERISK-22356)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2973/
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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/
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A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
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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/
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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
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
* Added an option flags parameter to interval hooks. Interval hooks now
can specify if the callback will affect the media path or not.
* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.
* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.
* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.
* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep. The agent entertainment is now changed from MOH to silence after
the alert beep.
* Fixed holding bridge technology to defer starting the entertainment. It
was previously a mixture of immediate and deferred.
* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.
* Miscellaneous holding bridge technology rework coding improvements.
Review: https://reviewboard.asterisk.org/r/2761/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a blond transfer (attended transfer that is completed
before the transfer recipient picks up) externally through chan_sip
or chan_pjsip would result in lost references to the channels
involved with the transfer as well as their bridge.
(closes issue ASTERISK-22092)
Reported by: mmichelson
Review: https://reviewboard.asterisk.org/r/2766/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP/foo -- Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar
Kick a ;1 channel and the chain toward SIP/foo goes away.
Kick a ;2 channel and the chain toward SIP/bar goes away.
This can leave a local channel chain between the kicked ;1 and ;2 channels
that are orphaned until you manually request one of those channels to
hangup or request the bridge to dissolve.
* Added ast_bridge_kick() as a companion to ast_bridge_remove(). The
functional difference is that ast_bridge_kick() may dissolve the bridge as
a result of the channel leaving the bridge.
* Made CLI "bridge kick <bridge> <channel>" use ast_bridge_kick() instead
of ast_bridge_remove() so the bridge can dissolve if needed.
* Renamed bridge_channel_handle_hangup() to ast_bridge_channel_kick() and
made it accessible to other files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The feature_attended_transfer test is failing due to Asterisk not
passing DTMF in the bridges created for internal attended transfers.
This sets the features initialization routine to set this flag by
default and adjusts the basic bridge and confbridge's use of the
bridging system accordingly as per Richard's suggestion instead of
adjusting this individual case. This change allows the necessary DTMF
to pass through the attended transfer bridge and complete the test
successfully.
Review: https://reviewboard.asterisk.org/r/2759/
(closes issue ASTERISK-22222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These problems were all caught by a test in the Asterisk Test Suite that
originated some Local channels and attempted to move the ;2 half of the Local
channel into a bridge using the Bridge AMI action.
(1) When originating a channel, the Newchannel event is emitted quickly;
however, the ;2 channel will not have a pbx thread assigned to it until
after the outbound 'dialing' for the ;1 is complete. Thus, there is a period
of time where the outside world "knows" of the channel's existence and can
influence it but Asterisk has not yet started the dialplan execution thread.
If a Bridge AMI action is taken on the channel, the channel appears to be a
Dialed channel with no PBX thread; hence, the channel will be imparted into
the Bridge by first 'yanking' the channel. At the same time, a race condition
can occur after the yank (but before entering the bridge) when ;1 answers
and starts a PBX on the ;2. The end result currently is an assertion failure
in the Bridging API, as a channel with a PBX is imparted into the Bridge.
There's no way to prevent AMI from attempting to Bridge a channel
immediately after creation; likewise, holding the channel lock through the
entire Dial operation is unwise (and impossible). Instead of treating the
presence of a PBX thread as an error, we simply bail out of the adding the
channel to the bridge through ast_bridge_impart. The Bridge action will
then fail - but we avoid a situation where the channel is both executing
a PBX thread and simultaneously being given a separate thread in the
bridging system (which would be a "bad thing"). Since imparting a channel
with a PBX *can* occur and is not a programming error, the asserts have been
removed.
(2) When the first condition occurs, we have to take one of two actions: either
hangup the yanked channel as it did not enter the bridge, or deref it
because we don't own it. We can determine if we own it or not by testing
for the presence of the PBX thread. If we hung it up directly, we'd crash.
(3) bridge_find_channel does not increase the reference count of the
ast_bridge_channel object. The RAII_VAR usage in ast_bridge_add_channel
thus created a ticking time bomb in whatever bridge the channel moved into,
as the destructor for the ast_bridge_channel object would be called.
Review: https://reviewboard.asterisk.org/r/2741/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents unreal channel optimization during the prequalification
phase when either channel is involved in DTMF emulation. This prevents
a situation where an emulated digit would be missed because the
emulation was never completed.
Review: https://reviewboard.asterisk.org/r/2747/
(closes issue ASTERISK-22214)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
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
Joshua Colp
Richard Mudgett
Kinsey Moore
Mark Michelson
Matthew Jordan
Jonathan Rose
Scott Griepentrog
Kevin Harwell
David M. Lee