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Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
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The approach with having a single global subscription to all extension
state changes has one issue: dynamically created hints don't have any
watchers and are therefore garbage collected on the first dialplan
reload.
This change creates a state subscription for every queue member with a
hint as state_interface, thus increasing the count of watches for
hints, so they are not destroyed prematurely anymore.
There are 2 side effects:
1. The state change callback in app_queue is not executed when
there are no members referring to the extension.
2. The callback is called multiple times for the same hint if it's
associated with more than one queue member.
Reported by: Steven T. Wheeler
ASTERISK-18411 #close
Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89
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ASTERISK-27301 #close
Change-Id: Ic31361f34e2de3b6470e68fc37205a7711082eba
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The previous patch for ASTERISK-27216 made it so you wouldn't get any
position or periodic announcements unless you had announce-to-first-user
enabled. The announce-to-first-user feature was added by ASTERISK_21782
as a result of the patch which introduced the redundant announcements that
ASTERISK-27216 removes.
* By noting that the makeannouncement variable is used to suppresses the
first user announcement, we set its initial value to the
announce-to-first-user enable setting.
ASTERISK-27216
Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a
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This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:
The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.
Reported-by: Stefan Engström
Tested-by: Stefan Engström
ASTERISK-27216 #close
Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0
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ASTERISK-19103 #close
Reported by: Jim Van Meggelen
Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b
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Fixed to use correct initial value and fixed to use the
correct queue info to check the first value.
ASTERISK-27204
Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73
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Add priority to callers in AMI QueueStatus response
ASTERISK-27092 #close
Change-Id: I8d1f737a72c7c38f4cfe1a4ee3ecc0a4f85bd199
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GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
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The fix for ASTERISK-25665 introduced a regression.
The return value of queue_exec used to be 0 in case of leavewhenempty
but it was changed to -1 (returned from wait_our_turn and passed
transparently by queue_exec), thus leading to hangup instead of returning
back to dialplan.
This commit resets the value back to 0 in this case, restoring
original behavior.
ASTERISK-27065 #close
Reported by: Marek Cervenka
Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac
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A change was done which added an 'in_call' flag to queue
members that was set to true while talking to an agent.
Unfortunately in practice this does not accurately reflect
whether they are talking to an agent or not. If a Local
channel is involved and a transfer is performed then the
app_queue application would incorrectly think the agent
was still in a call with the caller. This was done to
fix a race condition between an agent becoming available
by device state and the checking of the last call information
for the wrapup time. There was a small window where the
last call information would be the previous value instead
of the new one.
This change goes about fixing the original issue in a
different way by considering the call completed if device
state is received which would make the agent available
and if they are currently in a call. If this occurs the
last call information is updated before the agent becomes
available ensuring that old information is not present
when checking if the member should be called. This also
improves the transfer situation by actually updating
and enforcing the wrapup time.
ASTERISK-26399
ASTERISK-26400
ASTERISK-26715
ASTERISK-26975
Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea
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There are 2 places in app_queue.c that log EXITEMPTY event: one in
wait_our_turn, and another one in queue_exec in the loop trying to
call an agent after wait_our_turn.
In most cases it leads to logging EXITEMPTY twice.
ABANDON is also logged on two places, and in the rare case when an agent
and caller hang up simultaneously it's also possible to get duplicates
in queue_log.
This commit changes wait_our_turn to return -1 ("the caller should exit
the queue") instead of 0 ("the caller's turn has arrived") in case of
leaving when empty, so queue_exec skips the agent calling loop.
Also, leave_queue is now executed only once in this case, because 2nd
time is just a noop when the queue entry has already been removed.
Also, it sets qe->handled to -1 to indicate that the call was not
answered by an agent, but the necessary handling has already been done
in order to avoid logging an extra ABANDON entry.
ASTERISK-25665 #close
Reported by: Ove Aursand
Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e
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In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
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bridge"" into 13
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from queue" into 13
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This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27.
Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b
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Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.
This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.
ASTERISK-26862 #close
Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727
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The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.
Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088
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A caller can leave the Queue() application after being bridged with a
member in a few ways:
* Caller or member hangup
* Caller is transferred somewhere else (blind or atx)
* Caller is externally redirected elsewhere
The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.
This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.
ASTERISK-26400 #close
Reported by: Etienne Lessard
Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334
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ast_load_realtime_multientry() returns an ast_config structure whose
ast_categorys are keyed with the empty strings. Several modules were
giving semantic meaning to the category names causing problems at
runtime.
* app_directory: Treated the category name as the mailbox name, and
would fail to direct calls to the appropriate extension after an
entry was chosen.
* app_queue: Queues, queue members, and queue rules were all affected
and needed to be updated.
* pbx_realtime: Pattern matching would never succeed because the
extension entered by the user was always compared to the empty
string.
Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7
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With 500+ queues and a reload every minute, a random queue disappears
upon reload. The cause is mususe of the 'dead' flag. Namely, all queues
were marked dead up front, and then "resurrected" by dropping this flag
for those found in the configuration. But a queue marked dead can be
removed also when control leaves the app entry point on a PBX thread.
With this change, the queue is marked only not found, and at the end of
reload only the queues that are still not found are actually marked as
dead, so the dead flag is never reset, and set only on positively dead
queues.
ASTERISK-26755
Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf
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In some cases member is added to pending_members, and the channel
is hung up before any extension state change. So the member would
stay in pending_members forever. So when we call do_hang, we
should also remove member from pending.
ASTERISK-26621 #close
Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54
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sets the variable ABANDONED to TRUE if the call was not answered.
ASTERISK-26558
Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3
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When using Asterisk Realtime Architecture, empty fields are skipped and the
default values are used. If the "context" parameter in queue was set and then
cleared from the database, the old value remains in memory and it continues
to be used. This change initialize the "context" parameter with an empty value,
allowing clearing the parameter.
ASTERISK-26462 #close
Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905
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Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system. In this case string values
from a channel driver's peer and not from the user setting channel
variables.
* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
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The pause reason is not always cleared when it should be cleared.
* Made set_queue_member_pause() always clear pause reason if not pausing
with a reason string.
Change-Id: I993dad19626ec017478a230e980989438b778c53
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The output of CLI "queue show" and AMI Queues action is truncated and
"failed to extend from 240 to 327" messages are generated if the queue
member and interface names are lengthy.
* Increase the string buffer size from 240 to 512 in order to accommodate
for more information fields added to the output since v1.8.
ASTERISK-26360 #close
Reported by: Richard Mudgett
Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d
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When dialing channels it is possible that they may not ever
leave the not in use state (Local channels in particular) by
the time we cancel them. If this occurs but we know they were
dialed we explicitly remove them from the pending members
container so that subsequent call attempts occur.
ASTERISK-26299 #close
Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65
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When a call forward attempt is made from a Queue member, the current
code will hang up the forwarding channel in an off-nominal condition
prior to raising the Stasis events informing the rest of Asterisk that
the call was forwarded. This will result in a slew of dreaded FRACKs,
most likely leading to a crash.
This patch modifies the code such that we don't hang up the forwarding
channel even in an off-nominal condition until we've safely raised the
Stasis messages.
ASTERISK-25797 #close
Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38
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It is possible for a not in use state change to occur multiple
times causing a queue member to be removed from the pending call
container prematurely.
The first not in use state change will remove the queue member
from the container. At this moment the member may be called and
placed in the pending container. After this another not in use
state change can be received which will remove it from the
container. Despite being called at this point the code will
incorrectly see that there are no pending calls to it.
This change only removes it from the pending container if the
state has actually changed.
ASTERISK-26133 #close
patches:
app_queue.diff submitted by Richard Miller (license 5685)
Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0
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Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
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When unloading the app_queue module the members in each queue are
destroyed and as part of this they are removed from the pending
members container. Unfortunately a crash would occur as the container
was destroyed before the members were removed.
This change tweaks ordering so the container destruction occurs
after the members are destroyed.
ASTERISK-16115
Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b
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It was possible for a queue member that is a member of at least 2 or more
queues to receive mulitiple calls at the same time. This happened because
of a race between when a member was being rung and when the device state
notified the other queue(s) member object of the state change.
This patch makes it so when a queue member is being rung it gets added to
a global pool of queue members. If that same member is tried again, e.g.
from another queue, and it is found to already exist in the pending member
container then it will not ring that member.
ASTERISK-16115 #close
Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48
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ASTERISK-25954 #close
Reported by: Javier Acosta
Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256
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ASTERISK-25888 #close
Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117
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Fix calculate of average time for talktime is wrong when is completed the
first call beacuse the time for talked would be that call.
ASTERISK-25800 #close
Change-Id: I94f79028935913cd9174b090b52bb300b91b9492
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Member lastcall time is updated later than member status. There was chance to
check wrapuptime for available member with wrong (old) lastcall time.
New boolean flag "in_call" is set to true right before connecting call, and
reset to false after update of lastcall time. Members with "in_call" set to true
are treat as unavailable.
ASTERISK-19820 #close
Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500
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In app_queue added value Paused Reason on QueueMemberStatus when a member
on queue is paused and the reason was set.
ASTERISK-25480 #close
Reporte by: Rodrigo Ramírez Norambuena
Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e
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If a call enters on a queue and the members on that queue are updated in
realtime (ex: using mysql inserting a new agent) the queue members are
never refreshed and the call will stay in the queue until other event occurs.
This happens only if this is the first call of the queue and there is no
agents servicing.
This patch prevent this issue, ensuring realtime members are updated if
there is one call in the queue and no available agents
ASTERISK-25442 #close
Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682
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commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
refer ASTERISK-24958
above commit removed ast_channel_lock(qe->chan);
but failed to remove corresponding ast_channel_unlock(qe->chan);
ASTERISK-25561 #close
Reported Alec Davis
Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a
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* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel. This is usually
because of a call pickup.
Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.
ASTERISK-25423 #close
Reported by: John Hardin
Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172
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Replace inlined code with update_connected_line_from_peer().
ASTERISK-25423
Reported by: John Hardin
Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3
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When a queued caller transfers an agent to another extension sometimes the
raised AgentComplete event has a reason of "caller" and sometimes "transfer".
Since a transfer has taken place this should always be transfer. This occurs
because sometimes the stasis hangup event arrives before the transfer event
thus writing a different reason out.
With this patch, when a hangup event is received during a transfer it will
check to see if the channel that is hanging up is part of a transfer. If so
it will return and let the subsequently received transfer event handler take
care of the cleanup.
ASTERISK-25399 #close
Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d
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During some transfer scenarios involving queues Asterisk would sometimes
crash when trying to obtain a channel snapshot (could happen on caller or
member channels). This occurred because the underlying channel had already
disappeared when trying to obtain the latest snapshot.
This patch adds a reference to both the member and caller channels that
extends to the lifetime of the queue'd call, thus making sure the channels
will always exist when retrieving the latest snapshots.
ASTERISK-25185 #close
Reported by: Etienne Lessard
Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128
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* Extract set_queue_member_pause() from set_member_paused() for simpler
and more consistent code.
* Extract set_queue_member_ringinuse() from
set_member_ringinuse_help_members() for simpler code.
Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306
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