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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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POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.
Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
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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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Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended
With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.
AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.
Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.
ASTERISK-25925 #close
Reported by Mark Michelson
Change-Id: I42cbec7730d84640a434d143a0d172a740995543
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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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ASTERISK-25954 #close
Reported by: Javier Acosta
Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256
(cherry picked from commit c0688a6398f27296ff849848a2e416e036d794e3)
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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-25888 #close
Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117
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This eliminates some casts that I made a note saying v10 and above
would no longer need them.
Better late than never :)
Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
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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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Add time when started a the last pause for a queue member for
QueueMemberStatus ami event.
Also show accumulate time in seconds when started a pause for a queue
member to CLI command 'queue show'.
ASTERISK-16394 #close
Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c
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Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c
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Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d
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When the Asterisk is restared is not preseved reason paused of members.
This patch fixed this cases, retain data on astdb and set when Asterisk
is started.
ASTERISK-25732 #close
Report by: Rodrigo Ramírez Norambuena
Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5
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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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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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Add value of pause reason when is paused on CLI command "queue show"
ASTERISK-25581 #close
Report by: Rodrigo Ramírez Norambuena
Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864
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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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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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* 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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Setting the 'paused' and 'ringinuse' options on a queue member using the
dialplan function QUEUE_MEMBER did not behave the same way as the
equivalent dialplan applications or AMI actions.
* Made queue_function_mem_write() call the set_member_paused() and
set_member_value() for the 'paused' and 'ringinuse' options respectively.
A beneficial side effect is that the queue name is now optional and sets
the value in all queues the interface is a member.
* Update QUEUE_MEMBER XML documentation.
* Fix error checking in QUEUE_MEMBER() write.
ASTERISK-25215 #close
Reported by: Lorne Gaetz
Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb
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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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Change-Id: I7294e13d27875851c2f4ef6818adba507509d224
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Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
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This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
parameters: position, original position and waiting time.
ASTERISK-25038 #close
Reported by: Etienne Lessard
Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618
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* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
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A potential problem that can arise is the following:
* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.
If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.
Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.
The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:
* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.
This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:
* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.
The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.
Address review feedback on gerrit.
* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c
ASTERISK-24958 #close
Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
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Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
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Although it only occurred once, a crash occurred when a queue attempted to
evaluate a queue penalty rule that appeared to have already been destroyed.
In many locations in app_queue, a test is done to see if qe->pr is NULL;
however, when we dispose of a queue's penalty rules, we don't set the pointer
to NULL after free'ing it. This patch does that to prevent any dangling
pointers from lingering on the queue object.
Review: https://reviewboard.asterisk.org/r/4522
ASTERISK-23319 #close
Reported by: Vadim
patches:
rb4552.patch submitted by Stefan Engström (License 6691)
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Merged revisions 434449 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
evaluate to 'true'. This patch changes the evaluation to use
ast_strlen_zero.
* app_queue:
- Fixed evaluation of qe->parent->monfmt, which always evaluates to
true. Instead, we just check to see if the dereferenced pointer
evaluates to true.
- Fixed evaluation of mem->state_interface, wrapping it with a call to
ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
Review: https://reviewboard.asterisk.org/r/4541
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4541.patch submitted by dkdegroot (License 6600)
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Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes several warnings caught by clang - in this case, usage of the
abs function on non-integer values. This patch uses labs and fabs, as
appropriate, in the various affected files.
Review: https://reviewboard.asterisk.org/r/4525
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4525.patch submitted by dkdegroot (License 6600)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:
* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* main/security_events.c: removed unused TIMESTAMP_STR_LEN
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states
Review: https://reviewboard.asterisk.org/r/4526
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4526.patch submitted by dkdegroot (License 6600)
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Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to
(long) when printing members of certain time structs.
Review: https://reviewboard.asterisk.org/r/4507
ASTERISK-24879 #close
Reported by: snuffy
Tested by: snuffy
patches:
openbsd-time64.diff uploaded by snuffy (License 5024)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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instead.
ASTERISK-24049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Follow-up issue to -r430435 from reviewboard review.
ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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From reviewboard:
"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?
The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."
The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.
Review: https://reviewboard.asterisk.org/r/4135
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Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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