When r382230 added an option to not denoise the MeetMe conference (if a user
had a channel whose format's sample rate changed frequently, for example),
the value added was the maximum allowed value for the constants that define
the options for MeetMe in 1.8. Not so in 11 - unfortunately, the option
CONFFLAG_DONT_DENOISE conflicts with CONFFLAG_INTROUESR_VMREC. This patch
fixes that, and also tweaks one of the way in which the constants was
declared for consistency.
Thanks to Tony Mountifield for pointing out the problem and solution.
(closes issue ASTERISK-22269)
Reported by: Tony Mountifield
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.
(closes issue ASTERISK-21980)
Reported by: Avraam David
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk. Worse, support for reloads did not exist at first
and was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle. Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.
This patch converts various SLA objects to be reference counted objects
using astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear while one
of the other threads is using them. However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.
Review: https://reviewboard.asterisk.org/r/2581/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@393929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The config framework options should not be registered multiple times.
Instead the configuration just needs to be reprocessed by the config
framework.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r386792, the ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the first caller
to continue receiving prompts while the agent is dialed, it has the side effect
of preventing the first caller from hearing the agent immediately upon
bridging. This may not be a problem for those who really want this option, but
for those who didn't care whether or not the first caller in queue heard their
position, it was an issue.
This patch disables the ability for the first caller in the queue to hear
prompts and adds a new option, announce-to-first-user, to queues.conf. Those
who the behavior can enable it by setting this value to True.
Note that if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed.
(closes issue ASTERISK-21782)
Reported by: Remi Quezada
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The snapshot API contains an option that allow for combining of new
and old messages within a single snapshot. New messages, however,
include options beyond just 'INBOX' - it also includes the Urgent
folder. A previous patch that combined INBOX and Urgent accidentally
impacted snapshots that attempted to gain messages from just the Old
folder. This patch fixes the snapshot gathering such that the API
returns the appropriate messages for the folder selected, with and
without the combine option.
This should make it more clear about what's happening.
Review: https://reviewboard.asterisk.org/r/2539/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse". We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries. When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members. This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.
The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.
(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
asterisk-21738-rt-ringinuse-field-not-set.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2499/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reload support was originally not included for SLA. It was added later,
but in a fairly non-traditional way. It basically sets a flag
indicating that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then* executes
the reload. It does this because the reload process is destructive. It
starts by throwing everything away and starting over.
There are a number of problems with this approach. One of them is that
the check to see if anything in use was incomplete. This patch makes it
more complete and thus less likely for a crash to occur during reload
processing. However, this approach still has problems so some much more
significant reworking of this code will need to come in as a next step.
Patch credit and testing by CoreDial, LLC.
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When app_queue is unloaded, some manager commands are not being unregistered
which result in a segfault. This patch corrects this.
(closes issue ASTERISK-21397)
Reported by: Peter Katzmann, Corey Farrell
Tested by: Corey Farrell
Patches:
asterisk-21397-missing-unreg-manager-cmd_1.8.diff
Michael L. Young (license 5026)
asterisk-21397-missing-unreg-manager-cmd_11.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2444/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report was that app_voicemail would crash. This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status. After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.
During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.
This patch does the following:
* Creates a helper function to check if the configuration is valid
* Adds calls to the new helper function where appropiate
* Fixes memory leaks where the code returned without running
ast_config_destroy() on the configuration that was loaded
(closes issue ASTERISK-21302)
Reported by: Jaco Kroon
Tested by: Jaco Kroon, Michael L. Young
Patches:
asterisk-11.3.0-app_voicemail-ast_config-fixes.patch
Jaco Kroon (license 5671)
asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2443/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.
(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk. If more than
one station answers the call at the same time, all channels other than
the first one to answer are left in a bad state. The channel gets
leaked, is not connected to anything, and there's no way to get rid of
it.
We now properly clean up these losing channels by hanging up on them.
Since they lost the race, as we process their answer, there is no
ringing trunk for them to answer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A regression was accidentally introduced when allowing an optional ID to be used
when calling StopMixMonitor. When we are unable to stop MixMonitor on a
channel, -1 is being returned which triggers the hangup of the channel.
This patch restores the prior behavior by returning 0 whether we were successful
or not. It also allows the call from the manager to use the return code when
the action fails.
(closes issue ASTERISK-21294)
Reported by: daroz
Tested by: daroz
Patches:
asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2404/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.
It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fix checks to make sure that if a confbridge record start command is issued
from the CLI it will always use the file name given on the CLI even if it
changes between start/stop records for a conference. Previously it had been
reusing the same file between start/stops even if a new filename was given.
(issue AST-1088)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.
The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.
While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.
This patches does the following:
* Checks for the presence of func_speex as opposed to codec_speex when
determining if the DENOISE function is present (which is where the function
is actually implemented)
* Adds an option to MeetMe 'n' that causes the denoiser to not be applied
to a channel when it joins. This keeps the current behavior the default, but
let's users disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358
(closes issue AST-1062)
Reported by: Thomas Arimont
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ConfBridge was refactored to better handle the concept of marked,
wait_marked, and normal users co-existing in a conference (thereby implementing
a state machine for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active users. This list
is used for wait_marked users when they are waiting in a conference but no
marked user has joined; normal users may have joined at this point however.
There are several AMI/CLI commands that affect conference users that were not
checking the wait_marked users list:
* CLI/AMI commands that mute/unmute a participant. In this case, wait_marked
users have to remain in their particular state and should not be affected -
however, the commands would return "Channel not found" as opposed to the
appropriate error condition.
* CLI/AMI commands that kick a participant. An admin should always be able to
kick a participant out of the conference.
This patch fixes both sets of commands, and cleans up the CLI commands slightly
by allowing them to complete a participant name (this was supposed to have been
added, but the function call was commented out and wasn't implemented).
Review: https://reviewboard.asterisk.org/r/2346/
(closes issue AST-1114)
Reported by: John Bigelow
Tested by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.
This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.
Review: https://reviewboard.asterisk.org/r/2356/
(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:
return_context
The goto-style label to jump the call back into after timeout. Default
'priority+1'.
Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.
(closes issue ASTERISK-20113)
Reported by: serginuez
patches:
app_parkandannounce.diff uploaded by serginuez (License 6405)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Apparently this feature became broken in 11, probably as a result
of the Hangup Cause project.
(closes issue ASTERISK-21113)
Reprted by: Heiko Wundram
Patches:
app_dial.patch uploaded by Heiko Wundram (license 5822)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The vm_mailbox_snapshot_create function has an option that combines the
contents of INBOX and Old into a single snapshot. The intent of this is that
both 'new' messages and 'deleted' messages are given in a single snapshot, as
some applications prefer this view of the voicemail world. Unfortunately, the
initial implementation ignored the "Urgent" folder. The "Urgent" folder is a
pseudo-INBOX, in that new messages left with the 'U' flag will be placed in
that folder as opposed to INBOX. Thus, the option failed the intent with which
it was added.
This patch makes it so that the "Urgent" folder is included in the snapshot
when that option is used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you started/stopped recording of a conference multiple times channels
would remain active even when all participants left the conference. This
was due to the fact that a reference to the confbridge was being added
every time a start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the conference alive.
Made sure only a single reference is added for the record thread no matter how
many times recording is started/stopped. A de-reference is issued upon thread
ending.
Note, this issue is being fixed under AST-1088 since it relates to it and
should have been corrected along with those modifications.
(issue AST-1088)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock occurred after starting/stopping and then restarting a confbridge
recording. Upon starting a recording a record thread is created that holds a
lock until just before exiting. Stopping the recording does not stop/exit the
thread or release the lock. The thread waits until recording begins again.
Starting a stopped recording signals the thread to continue and start recording
again. However restarting the recording also created another record thread
resulting in a deadlock. The fix was to make sure the record thread was only
created once.
Also it was noted that filenames for the recordings were being concatenated for
each start/stop. This was fixed by creating a new file for each conference
session and appending the actual recorded data within the file (e.g. passing
the 'a' option to MixMonitor).
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function call ast_db_deltree returns the number of row deleted, or a
negative number if it failed. DBdeltree was treating any non-zero return
as an error, causing a spurious verbose error message to be displayed.
This patch handles the return code of ast_db_deltree correctly.
(closes issue ASTERISK-21070)
Reported by: ianc
patches:
dbdeltree.diff uploaded by ianc (License #5955)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If say.conf did not exists prior to originally loading module app_playback it
would not load on subsequent reloads of the module once it had been created.
This occurred because upon reload of the app_playback module it would only
load a new configuration if an old one had previously existed. This fix simply
removed the association between checking if an old configuration existed and
the loading of the new one.
(closes issue ASTERISK-20800)
Reported by: pgoergler
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unloading ConfBridge caused the next AMI action received to crash
Asterisk.
* Add the missing unregister of AMI action ConfbridgeSetSingleVideoSrc
when ConfBridge is unloaded.
(closes issue ASTERISK-20994)
Reported by: Jeremy Kister
Patches:
jira_asterisk_20994_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: Rusty Newton, Jeremy Kister
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Page and ConfBridge custom announcement did not play when users
entered the conference.
* Fix the CONFBRIDGE(user,announcement) file not getting played. The code
to do this got removed accidentally when the ConfBridge code was
restructured to be more state machine like.
* Fixed play_prompt_to_user() doxygen comments.
* Fixed the Page A(x) and n options for the caller. The caller never
played the announcement file and totally ignored the n option. The code
to do this was lost when the application was converted to use ConfBridge.
* Factored out setup_profile_bridge(), setup_profile_paged(), and
setup_profile_caller() routines to setup ConfBridge profiles. Made each
profile setup routine use the default template if one has not already been
setup by dialplan.
(closes issue ASTERISK-20990)
Reported by: Jeremy Kister
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A marked user ending a conference with only end_marked users generates
error messages:
ERROR[0000][C-00000000]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user ''
* The MULTI_MARKED state was doing too much when it was kicking out the
end_marked users from the conference. The kicked out users will clean up
after themselves when they exit the conference.
(closes issue ASTERISK-20991)
Reported by: Jeremy Kister
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "sound_only_one" sound was not being set even though it was configured. In
looking into this, I found that the "join" and "leave" prompts were not being
set either.
(closes issue ASTERISK-20898)
Reported by: Stephan
Tested by: Stephan
Patches:
asterisk-20898-custom-sounds-ignored.diff uploaded by
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2289/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for ConfbridgeList states that the Conference field is
optional. That's not really the case: if you fail to provide a Conference
number, the command will kick back an error.
(closes issue AST-1090)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The old prompts for the administrator menu were inadequate. They didn't mention
that the menu had additional options through the 8 key and pressing the 8 key
wouldn't reveal what those options were. This patch fixes all of that while
also organizing code pertaining to each individual menu type which was
previously all stored in one gigantic function along with many of the basic
conference functions.
(closes issue AST-996)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/360/
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This patch fixes two bugs:
* If an outbound call is made from a SLA phone using SLAStation, then there is
no ringtone audible to the phone that originates the call. The indication of
the ringing was not being passed to the SLA station; this patch fixes that
by passing through the progress indications.
* If an SLA station hangs up before the called party answers, then the channel
to the called party continues to ring until a timeout occurs. If the called
party manages to answer, Asterisk attempts to connect the called party to
a non-existant MeetMe room. This patch corrects the behavior by abandoning
the call attempt if it detects that the SLA station is no longer in use
while attempting to call the called party.
Review: https://reviewboard.asterisk.org/r/2275/
(closes issue ASTERISK-20462)
Reported by: dkerr
patches:
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20462.patch uploaded by dkerr (license 5558)
(closes issue ASTERISK-20440)
Reported by: dkerr
patches:
asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
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* Generate a warning message if sound files do not exist when trying to
play the user count to the conference. Use the new helper routine
sound_file_exists() for consistency.
* Put the new user into autoservice when playing user counts to the
conference.
* Check the return value of ast_bridge_impart().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An incorrect string initializations was left in ast_str_encode_mime from the
patch that converted string manipulations to use ast_str strings (r191140).
The string initialization causes a crash when ast_str_set is called on
the string later on in the function.
(closes issue ASTERISK-18697)
Reported by: Chris Boot
patches:
minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309)
(issue ASTERISK-20854)
Reported by: Chris Warr
Tested by: Chris Warr
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When the restructuring work got committed to Confbridge in r375470 to
fix many open issues, it caused a regression in the reported count of
users when conference information was requested via CLI or manager.
This corrects the user count and user information displayed when
listing conference information from the CLI and manager.
(closes issue ASTERISK-20938)
Reported By: Timo Teras
Patches:
confbridge-list.patch uploaded by Timo Teras (license 5409)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_voicemail will no longer issue error messages when it retrieves an msg_id
with a NULL value from realtime and will instead simply populate the msg_id
field with a newly generated msg_id. In addition, this patch changes the way
msg_ids are generated to eliminate certain causes of duplicate IDs appearing
within a single system. In addition, when messages are copied, they will now
receive a new msg_id.
(closes issue ASTERISK-20717)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/2220/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from a queue.
NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.
* Did some refactoring to eliminate some code redundancy.
(issue ASTERISK-16115)
Reported by: nik600
Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
Modified
* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem. The fix did not need to be optional. The fix should not have
tried to explicitly set the device state. Setting the device state by
something other than the device introduces a race condition. I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.
This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not. It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches:
asterisk-20743-q-cmplt-caller.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2256/
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Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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With the option ringinuse=no set, the patch committed for ASTERISK-16115
causes non-SIP queue members to never be called because the device state
is checked after a channel is created to determine if the member is busy.
These queue members always get the "Member %s is busy, cannot dial"
message.
Most channel drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or unknown
if the channel exists or not respectively.
(closes issue ASTERISK-20801)
Reported by: rmudgett
Patches:
jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett
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When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
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* Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t.
* Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so
destroy_conference_bridge() can destroy them unconditionally.
* Made join_conference_bridge() abort if the new conference could not be
added to the conferences container.
* Made leave_conference() discard any post-join actions if
join_conference_bridge() had to abort early.
* Made the join_conference_bridge() diagnostic messages better describe
what happened.
* Renamed leave_conference_bridge() to leave_conference() and made it only
take a conference user pointer. The conference pointer was redundant.
* Made conf_bridge_profile_copy() use struct copy instead of memcpy().
* No need to lock the conference in start_conf_record_thread() since all
of the callers already have it locked.
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* Made func_confbridge_helper() allow an empty value when setting options.
You previously could not Set(CONFBRIDGE(user,pin)=) and clear the
configured pin from the dialplan.
* Made func_confbridge_helper() handle its datastore better if multiple
threads attempt to set the first CONFBRIDGE option value on the channel.
* Made the func_confbridge_helper() only output one diagnostic message
concerning the option.
* Made the bridge video_mode able to repeatedly change in the config file
and CONFBRIDGE dialplan function. The video_mode option values are an
enum and not independent of each other.
* Made handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option.
* Simplified datastore handling code in conf_find_user_profile() and
conf_find_bridge_profile().
(closes issue ASTERISK-20655)
Reported by: Birger "WIMPy" Harzenetter
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* Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
it is effectively a noop. No channels can attach a reference to that
module.
* Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
The caller of unload_module() has already called it.
* Removed redundant channel module references in pbx_dundi.c. The
registered dialplan function callback dispatchers for the read/read2/write
callbacks already reference the module before calling.
* pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
functions to the first thing the unload_module() does. This will reduce
the chance of new channels using DUNDi services while the module is being
torn down.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a test event to ConfBridge that reports transitions between
states in ConfBridge. This is used by tests in the Asterisk Test Suite
that verify state changes based on the entering/leaving of conference
participants.
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Channels would get stuck and MeetMe would repeatedly display an Unable
to write frame to channel error in the conf_run function if hung up
during certain sound prompts such as during user count announcements.
This patch fixes that by reintroducing a hangup check in the meetme's
main loop (also in conf_run).
(closes issue ASTERISK-20486)
Reported by: Michael Cargile
Review: https://reviewboard.asterisk.org/r/2187/
Patches:
meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
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We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists
and is the correct sound file. Previous behavior was silence and a warning on the CLI.
(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
asterisk20280.patch uploaded by Rusty Newton (license 5829)
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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When confbridge was changed to handle conference status with a state machine in
r374658. The function responsible for starting recording for a conference was
refactored with the function actually responsible for launching the recording
thread being split into a function with another name. The old function name was
still used for manually started recordings through AMI or CLI. This patch fixes
that by switching which function is used to start recording the conference.
(closes issue ASTERISK-20601)
Reported by: Vilius
Patches:
confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182)
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Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
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If a format name was not found by ast_getformatbyname, a NULL pointer
would be passed into ast_format_rate and immediately dereferenced.
This ensures that a valid pointer is used since the structure is
already allocated on the stack.
(closes issue DPH-523)
Reported-by: Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Party A calls Party B
Party B puts Party A on hold.
Party B calls a queue.
Ringing queue member D sees Party B identification.
Party B transfers Party A to the queue.
Queue member D does not get a connected line update for Party A.
Queue member D answers the call and still sees Party B information.
However, if Party A later transfers the call to Party C then queue member
D gets a connected line update for Party C.
* Made pass connected line updates from the caller to queue members while
the queue members are ringing.
(closes issue AST-1017)
Reported by: Thomas Arimont
(closes issue ABE-2886)
Reported by: Thomas Arimont
Tested by: rmudgett
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Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches
it became clear that app_confbridge had some complex logic in how it handled
interactions between marked, waitmarked, and unmarked users. In particular,
there were some areas in which the interactions between the users resulted
in inconsistent behavior, and app_confbridge was missing logic in how to handle
some corner cases. Some areas included:
* Poor handling of mixing unmarked and waitmarked users
* Inconsistencies in how MOH and muting was applied to various users
* Handling of various announcements for different user profile options
flan's patches seem to fix the various issues, but highlighted how hard the
code could be to maintain. In an attempt to make things easier to maintain and
to more fully enumerate the various cases that exist, this patch breaks up the
logic into a state machine-like setup.
Please note that the various state transitioned are documented on the Asterisk
wiki:
https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
Review: //https://reviewboard.asterisk.org/r/2072/
Note that for the following issues, mjordan uploaded the patch, although it
was written by twilson. Any contributor license discrepency is due to that.
(closes issue ASTERISK-19562)
Reported by: flan
Tested by: flan, mjordan, jrose
patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)
(closes issue ASTERISK-19726)
Reported by: flan
Tested by: flan
patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)
(closes issue ASTERISK-20181)
Reported by: Jonathan White
Tested by: Jonathan White
patches:
bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)
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Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case. This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.
The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.
As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.
Review: https://reviewboard.asterisk.org/r/2136/
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Not panicking means that the old config is kept.
(closes issue ASTERISK-20458)
Reported by: Leif Madsen
Patches:
ASTERISK-20458.patch uploaded by Mark Michelson(license #5049)
Tested by Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix previously untested senarios;
1). On queue initialisation set queue_avail devstate to INUSE.
Previously was unavailable, which indicated an agent was available.
2). When removing members, if there are no other members available, set queue_avail to INUSE.
Previously, if a member interface had become 'unavailable', they were never going to be removed, particularly when persistant queues is enabled.
3). When adding a member, check that they are available, if they are set queue_avail to NOT_INUSE.
Previously on reloaded, members may have been 'unavailable'.
4). When pausing or unpausing a member, set appropriate queue availability.
alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2129/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see
if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one.
This operation left around a pointer to the last created conference bridge still containing participants.
When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of
and the reference count of the conference bridge decremented. If there was only a single participant in the
conference bridge it was ultimately destroyed prematurely.
(closes issue AST-994)
Reported by: John Bigelow
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Sets INUSE when no free agents, NOT_INUSE when an agent is free.
modifes handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate.
Previously exited early if the member was found in the queue.
Now Exits later when both a member was found, and a free agent was found.
alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2121/
~~~~
Support all ways a member can be available for 'agent available' hints
Alec's patch in r373188 added the ability to subscribe to a hint for when
Queue members are available. This patch modifies the check that determines
when a Queue member is available by refactoring the availability checks in
num_available_members into a shared function is_member_available. This
should now handle the ringinuse option, as well as device state values
other than AST_DEVICE_NOT_INUSE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for hints on a queue. Hints can be added using
the nomenclature 'Queue:name', where name is the name of the queue being
monitored.
This nifty feature was done by Alec Davis.
Review: https://reviewboard.asterisk.org/r/1619
Reported by: Alec Davis
Tested by: alecdavis
patches:
review1619.diff2 by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* ASTERISK-20383
Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.
* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up. In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.
Regression because of the named call pickup group feature.
* See ASTERISK-20386 for the implementation improvements. These are the
changes in channel.c and channel.h.
* Fixed some locking issues in CHANNEL().
(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2112/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI action VoicemailUsersList VoicemailUserEntry event headers
ServerEmail and MailCommand did not report the global values if they were
not overridden. The VoicemailUserEntry event header ServerEmail was not
populated with the global value if the voicemail user did not override it.
The VoicemailUserEntry event header MailCommand was never populated with a
value.
* Removed unused struct ast_vm_user member mailcmd[].
(closes issue AST-973)
Reported by: John Bigelow
Tested by: rmudgett
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When MiniVM sends an e-mail and it has the volgain option set, it will spawn
sox in a separate process to handle the manipulation of the sound file. In
doing so, it creates a temporary file. There are two problems here:
1) The file descriptor returned from mkstemp is leaked
2) The finalfilename character pointer points to a buffer that loses scope
once volgain processing is finished.
Note that in r316265, Russell fixed some gcc warnings by using the return
value of the mkstemp call. A warning was placed in minivm that the file
descriptor was going to be leaked. This patch reverts that change, as it
handles the leak and 'uses' the file descriptor returned from mkstemp.
(closes issue ASTERISK-17133)
Reported by: Tzafrir Cohen
patches:
minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035)
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The Status: header in a QueueMemberStatus event (and other QueueMember* events)
is the numeric value of the device state corresponding to that Queue Member.
As those values are not exactly obvious, listing them in the documentation is
useful.
Matt Riddell reported this indirectly through the wiki page.
(closes issue ASTERISK-20243)
Reported by: Matt Riddell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using tab-completion for the list of queues on "queue reset stats"
or "queue reload {all|members|parameters|rules}", the tab-completion
listing for further queues erroneously listed queues that had already
been added to the list. The tab-completion listing now only displays
queues that are not already in the list.
(closes issue AST-963)
Reported-by: John Bigelow
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Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.
(closes issue AST-958)
Reported-by: John Bigelow
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When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters. This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the configuration
file.
Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256
characters. Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.
(closes issue ASTERISK-16879)
Reported by: Clod Patry
Tested by: mjordan
patches:
followme_no_limit.diff uploaded by Clod Patry (license #5138)
Slightly modified for this commit.
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This patch fixes two memory leaks:
1. When find_user is called with NULL as its first parameter, the voicemail
user returned is allocated on the heap. The inboxcount2 function uses
find_user in such a fashion when counting new messages, and fails to free
the resulting voicemail user object.
2. When populate_defaults is called on a voicemail user, it wipes whatever
flags have been set on the object by copying over the global flags object.
If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
that flag is removed. This leaks the voicemail user when free_user is later
called.
(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
asterisk.patch2 uploaded by Filip Jenicek (license 6277)
Patch slightly modified for this commit.
Review: https://reviewboard.asterisk.org/r/2096
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When app_queue is unloaded, the queues container has its refcount
decremented, potentially to 0. Then the taskprocessor responsible
for handling device state changes is unreferenced. If the
taskprocessor happens to be just about to run its task, then it
will create and destroy an iterator on the queues container.
This can cause the refcount on the queues container to increase to
1 and then back to 0. Going back to 0 a second time results in
double frees.
This failure was seen periodically in the testsuite when Asterisk
would shut down.
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Queue member status would not always get updated properly when the member
was called, thus resulting in the member getting multiple calls. With this
change, we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call before
placing an outbound call.
(closes issue ASTERISK-16115)
reported by nik600
Patches:
app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409)
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If a static queue had realtime members, then there could be a potential
for those realtime members not to be properly deleted from memory.
If the queue's members were loaded from realtime and then all the
members were deleted from the backend, then the queue would still
think these members existed. The reason was that there was a short-
circuit in code such that if there were no members found in the
backend, then the queue would not be updated to reflect this.
Note that this only affected static queues with realtime members.
Realtime queues with realtime members were unaffected by this issue.
(closes issue ASTERISK-19793)
reported by Marcus Haas
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* The RemoveQueueMember app made mention of options that could
be passed in, but no options are supported. I have removed the
listing of options from the documentation.
* The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
value that could be set.
(closes issue AST-949)
reported by Steve Pitts
(closes issue AST-954)
reported by Steve Pitts
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Matthew Jordan
Walter Doekes
Russell Bryant
Richard Mudgett
Jonathan Rose
Jason Parker
Michael L. Young
Olle Johansson
Joshua Colp
Kevin Harwell
Kinsey Moore
Rusty Newton
Mark Michelson
Sean Bright
Alec L Davis