Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
ASTERISK-28363
Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
Because the per-mailbox options are the last thing on a line, don't look
for or stomp on any subsequent commas.
ASTERISK-27935 #close
Reported by: Sébastien Duthil
Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153
If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.
ASTERISK-20207 #close
Reported by: Steven Wheeler
Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf
* Always set member->lastpause when setting member->paused
* Fixed typo (using member->lastcall instead of member->lastpause) in
'queue show' output.
* Use a constant 'now' in 'queue show' output for a better point-in-time
view of time based stats.
ASTERISK-27541 #close
Reported by: César Benjamín García Martínez
Change-Id: Ib41ced90cfdb66f9bb1e7b263d0f6fc1ac6e18fa
It was a copy/paste of the QUEUE_MEMBER_COUNT function's synopsis.
ASTERISK-20986 #close
Reported by: Olivier Krief
Change-Id: If51ec481feb35824a4e78ab5600b197b819b10be
Fixes an intermittent segmentation fault which occured when accessing
nativeformats of a channel which entered into a queue.
ASTERISK-27964
Reported by: Francisco Seratti
Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e
Topic names now follow: <subsystem>:<functionality>[/<object>]
This ensures that they are all unique, and also provides better
insight in to what each topic is for.
Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.
Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.
Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.
ASTERISK-28335
Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d
While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.
Also cleaned up an adjacent error message for consistency.
ASTERISK-28168 #close
Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202
This change add ability to set the wrapuptime per-member using the
AddQueueMember application.
The feature to set wrapuptime per member was include in the issue
ASTERISK-27483 for static member by configuration file and was not
added to set from AddQueueMember.
ASTERISK-28055 #close
Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf
When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.
This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.
This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.
ASTERISK-28277
Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6
You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes. These can be used anywhere
the mailbox is specified.
Example:
[general]
aliasescontext = myaliases
[default]
1234 = yadayada
[myaliases]
4321@devices = 1234@default
Now you can use 4321@devices to refer to the 1234@default mailbox.
This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.
Change-Id: I395b9205c91523a334fe971be0d1de4522067b04
If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is
not updated correctly since urgent messages are in a different directory. The
fix is to update the channel variable when the path to the urgent message is
created.
ASTERISK-28225
Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca
When using the 'b' option to Queue with a queue that was not configured
for ring all a crash would occur as the wrong pointer would be used.
ASTERISK-28218
Change-Id: If1390f64e321047dff24fd2410c95dde74904980
The free_user function was automatically deleting the stasis mailbox
state but this only makes sense when the mailbox is actually
deleted, not just the structure freed. This was causing issues
where leave_voicemail would publish the mwi message to stasis and
delete the state before the message could be processed by
res_pjsip_mwi.
* Removed the delete of state from free_user().
* Created a new free_user_final() function that both frees the data
structure and deletes the state. This function is only called
during module load/unload where it's appropriate to delete the
state.
ASTERISK-28215
Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd
Currently the file sound_only_person is not played when a marked
user (with announce_only_user=yes) joins an empty conference.
This patch fixes it.
ASTERISK-28201 #close
Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4
This reverts commit 29115e2384.
That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf. This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.
ASTERISK-28151
Reported by: Ronald Raikes
Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15
* The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
* The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
* A topic pool is now used for individual bridge topics.
* The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
* A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
* The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
* A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
* The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
* The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.
* cdr, cel, manager and ari have been updated to use the new
arrangement.
Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369
When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.
This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.
The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.
Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.
ASTERISK-28119
Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423
Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.
As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.
The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.
ASTERISK-28102
Change-Id: I9334febff60a82d7c39703e49059fa3a68825786
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
This replaces the inline functions with macros. This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.
Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.
This patch also blocks updates in the other direction before call is
answered.
ASTERISK-27980
Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01
Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel. Only bridge_softmix has that
data so now it's set when the bridge topology is changed.
ASTERISK-28107
Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499
* Update the post-answer documentation and example. The Dial example was
incorrect and misleading for the post-answer subroutine useage.
* Fix note and warning paragraphs in option descriptions. They don't show
up in the wiki.
Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
Declining the queue_member_status_type stasis message in stasis.conf
causes these messages to leak json objects.
* Add missing ast_json_unref() if the type is NULL in
queue_publish_member_blob().
ASTERISK-28084
Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc
The first attempt at publishing confbridge events to participants
involved publishing them at the same time stasis events were
created. This caused issues with bridge and channel locks. The
second attempt involved publishing them when the stasis events
were received by the code that published the confbridge AMI events.
This caused timing issues because, depending on resources available,
the event could be received before channels actually joined the
bridge and would therefore fail to send messages to the participant.
This attempt reverts to the original mechanism with one exception.
The join and leave events are published via bridge join and leave
hooks. This guarantees the states of the channels and bridge and
provides deterministic timing for event publishing.
Change-Id: I2660074f8a30a5224cb953d5e047ee84484a9036
This issue related to setting of holdtime, announcements, member delays.
It works well if we set the member delays to "0" and no announcements
and no holdtime.This issue will happen if we set member delays to "1",
"2"... or announcements or holdtime and hangs up the call during
processing it.
And here is the reason:
(At the step of answering a phone.)
It takes care any holdtime, announcements, member delays,
or other options after a call has been answered if it exists.
Normally, After the call has been aswered,
and we wait for the processing one of the cases of the member delays
or hold time or announcements finished, "if (ast_check_hangup(peer))"
will be not executed, then queue will be updated at update_queue().
Here, pending member will be removed.
However, after the call has been aswered,
if we hangs up the call during one of the cases of the member delays
or hold time or announcements, "if (ast_check_hangup(peer))"
will be executed.
outgoing = NULL and at hangupcalls, pending members will not be removed.
* This fixed patch will remove the pending member from container
before hanging up the call with outgoing is NULL.
ASTERISK-27920
Reported by: Cao Minh Hiep
Tested by: Cao Minh Hiep
Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855
app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload. This resulted in leaks in both
areas.
* app_voicemail now calls ast_delete_mwi_state_full when it frees
a user structure and ast_delete_mwi_state_full in turn now calls
the new stasis_topic_pool_delete_topic function to clear the topic
from the pool.
Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8
The append_mailbox function wasn't calculating the correct length
to pass to ast_alloca and it wasn't handling the case where context
might be empty.
Found by the Address Sanitizer.
Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161
app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers. It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled. For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.
Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.
This paves the way for disabling the caching of stasis subscription
change messages.
Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.
ASTERISK-27121
Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee
I'm only seeing an error in 14+, so I assume it is due to different
compiler options:
app_queue.c: In function ‘handle_queue_add_member’:
app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11
bytes into a region of size 3 [-Werror=format-overflow=]
sprintf(num, "%d", state);
^~
app_queue.c:10234:18: note: directive argument in the range
[-2147483648, 99]
sprintf(num, "%d", state);
^~~~
Compiler: gcc version 8.0.1 20180414 (experimental)
[trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2)
Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10
When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.
The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.
ASTERISK-27973 #close
Reported-by: Valentin Safonov
Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.
Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label"
attribute to the current sdp.
Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5
If a conference is ended very quickly after it was created (i.e., the
first user immediately hangs up) then the conference bridge and announcer
channels are not removed.
When a conference is created, the push_announcer() function is added to
the playback queue task processor and the conference object reference is
bumped. If a conference is ended while the push_announcer() function is
still going then the ao2_cleanup(conference) at the end of
push_announcer() will call the destructor function -
destroy_conference_bridge().
The destroy_conference_bridge() function will then add the
hangup_playback() task to the playback queue and will wait for it to end.
Since it is already a current task of the playback queue it will wait
forever.
This patch makes the conference thread call push_announcer() directly.
This way the conference object reference bump is not needed. Since the
playback queue task processor is only used by the conference thread
itself, there is no danger of trying to play announcements before the
announcer is pushed to the bridge.
ASTERISK-27870 #close
Change-Id: I947a50fb121422d90fd1816d643a54d75185a477
With the participant info code in app_confbridge, we were still
in the process of adding the channel to the bridge when trying to send
an in-dialog MESSAGE. This caused 2 threads to grab the channel
blocking flag at the same time. To mitigate this, the participant
info code was moved to confbridge_manager so it runs after all
channel/bridge actions have finished.
Change-Id: I228806ac153074f45e0b35d5236166e92e132abd
Add predial handler support to app_queue. app_dial (ASTERISK_19548) and
app_originate (ASTERISK_26587) have the ability to execute predial
handlers on caller and callee channels. This patch adds predial handlers
to app_queue and uses the same options as Dial and Originate (b and B).
The caller routine gets executed when the caller first enters the queue.
The callee routine gets executed for each queue member when they are about
to be called.
ASTERISK-27912
Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905
This patch changes the way asterisk polls output from mpg123, instead
of waiting for 10 seconds(when playing an http url) it now uses a
timeout of one second and iterates 10 times using this same timeout.
The main difference is that for every timeout asterisk receives it now
checks if mpg123 is still running before poll again.
ASTERISK-27752
Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620
ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc. In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.
For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.
To control this behavior, the following options have been added to
confbridge.conf:
bridge_profile/enable_events: This must be enabled on any bridge where
events are desired.
user_profile/send_events: This must be set for a user profile to send
events. Different user profiles connected to the same bridge can have
different settings. This allows admins to get events but not normal
users for instance.
user_profile/echo_events: In some cases, you might not want the user
triggering the event to get the event sent back to them. To prevent it,
set this to false.
A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used. This allows participant A's video
stream to appear as the same label to all other participants.
Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e
There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before. Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.
Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9
When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.
ASTERISK-27877 #close
Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6
The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and
MeetmeTalkRequest AMI events were documented with sending out a Usernum
header when the User header was actually output.
* Change the online documentation to match reality.
ASTERISK-27873
ASTERISK-25261
Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39
Fix data-type mismatch between app_voicemail and database columns
exposed by new version of MariaDB
ASTERISK-27760
Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b
Correct the log warning message shown when ODBC voicemail
retrieve_file is called and there is a null value in the category
column.
A more meaningfull message is now written at debug level.
ASTERISK-27853
Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4
This fixes build warnings found by GCC 8. In some cases format
truncation is intentional so the warning is just suppressed.
ASTERISK-27824 #close
Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84
Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will
prevent crash if dialplan contains an include loop. Log an error when
this occurs, at most one message per call to Macro() so we avoid logger
spam.
ASTERISK-26570 #close
Change-Id: I6c71b76998c31434391b150de055ae9a531e31da
Fixes a bug on the "confbridge show profile bridge" cli command
that showed "video_mode=no video" when video_mode was set
to "sfu"
ASTERISK-27418 #close
Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9
This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.
Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.
Support for configuring which behavior to use has been
added to app_confbridge.
ASTERISK-27804
Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66
SendText now accepts new channel variables that can be used
to override the To and From display names and set the Content-Type
of a message. Since you can now set Content-Type, other text/*
content types are now valid.
Change-Id: I648b4574478119f95de09d9f08e9595831b02830
This change adds a configuration option to app_confbridge which can be
used to set the interval at which we will send a combined REMB (remote
estimated maximum bitrate) frame to sources of video. The bridging API
has also been extended slightly to allow setting this so bridge_softmix
can use it.
ASTERISK-27786
Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82
Add an option to make app_originate not wait for the created channel
to answer.
Change-Id: I7fc2facd77079abc6321f44e8bcd4e39298de2ae
Requested-by: Frederic Steinfels <fst@highdefinition.ch>
Signed-off-by: Russell Bryant <russell@russellbryant.net>
Asterisk does not need the development package of libltdl, because it does not
use any symbol of -lltdl directly. Instead, it uses the runtime package via the
shared library -lodbc. On the supported platforms, that shared library declares
its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have
failed.
ASTERISK-27745
Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba
Certain applications (e.g. door-phone) require that also video is transmitted
before a call is accepted.
Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e
When app_voicemail calls ast_test_suite_notify with the results of
a user keypress, it formats the keypress as '%c'. If the user hung up
or some other error occurrs, the result of the keypress is a non
printable character. This ultimately causes json_vpack_ex to think
it's being passed a non utf-8 string and return an error.
* Keypress results passed to ast_test_suite_notify are now checked with
isprint() and a '?' is substituted if the check fails.
Change-Id: I78ee188916bbac840f3d03f40201b692347ea865
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
The menuselect comment was updated to deprecate these modules but the
AST_MODULE_INFO block at the end of file was missed.
ASTERISK-27671
Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
astosp.h is leftover from when logic was split between app_osplookup and
res_osp. All logic was moved into app_osplookup by 109737eb1c in 2006,
but astosp.h remained. This moves the remaining defines into
app_osplookup and deletes astosp.h.
Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd
Between Asterisk 11 and Asterisk 13 there was a significant increase
in the number of AST_FRAME_NULL frames being processed by app_amd.c's
main loop. Each AST_FRAME_NULL frame was being counted as 100ms
towards the total time and silence. This may have been accurate
when app_amd.c was orginally added, but it is not in Asterisk 13.
As such the total analysis time and silence calculations were way
off effectively breaking app_amd.c
* Additional debug messages were added
* AST_FRAME_NULL are now ignored
ASTERISK-27610
Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9
* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).
This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.
ASTERISK-27671 #close
Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49
* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers. The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped. However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".
ASTERISK-27651
Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
I've audited all modules that include any header which includes
asterisk/optional_api.h. All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.
In addition ARI dependency declarations have been reworked. Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.
Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606
This patch adds the ability to configure a prompt which will be read
to the "winner" who pressed 1 (or the configured value) and received
the call.
ASTERISK-24372 #close
Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
The check for last_user == NULL needs to happen before we dereference
the variable, previously it was possible for us to check flags of a NULL
last_user.
Change-Id: I274f737aa8af9d2d53e4a78cdd7ad57561003945
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
* mwi_sub_event_cb: mwist leaked on separate_mailbox failure.
* add_email_attachment: A reference to sox_gain_tmpdir was used
after the storage was out of scope.
Change-Id: I6282c542ff7b82fa091177a912d11234a8b00a30
This patch adds the ability to set the wrapuptime on the queue member
config.
When the option is set the wrapuptime on the queue member is used instead
of the queue's wrapuptime.
ASTERISK-27483 #close
Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902
The Local channel has never supported app_transfer
from what I can see so remove it from the documentation.
ASTERISK-25649
Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
Currently, to figure out specified voicemail's status, there's only one
way to do it, which is use a VoicemailUserEntry AMI message.
But it consumed it too much resource(it check everything).
So, added new AMI action.
ASTERISK-27470
Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7
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
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's
no OldMessageCount info for default.
To check the OldMessageCount info, it required IMAP_STORAGE define, but this is
not correct.
Added OldMessageCount item as a default.
ASTERISK-27456
Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7
Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.
Change-Id: I0123258eafce324249433a69df15a85cc16e509f
Declare 'res' initialized to -1 to deal with earlier error paths that
could cause 'res' to be returned uninitialized.
Change-Id: I8ac2a5755bf4174d89ef893e924c940f702b104e
We've been calling pbx_builtin_setvar_helper to set the
RECORD_STATUS variable before actually closing the recorded file.
If a client is watching VarSet events and tries to do something with
the file when a RECORD_STATUS event is seen, they might attempt to
do so while the file it's still open.
We now delay calling pbx_builtin_setvar_helper until after we close
the file.
ASTERISK-27423
Change-Id: I7fe9de99953e46b4bafa2b38cf151fe8f6488254
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23
* Stop using ast_module_helper to check if a module is loaded, use
ast_module_check instead (app_confbridge and app_meetme).
* Stop ast_module_helper from listing reload classes when needsreload
was not requested.
ASTERISK-27378
Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239
Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'
ASTERISK-27377 #close
Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930
* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.
ASTERISK-27350
Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc
We were ignoring the return value from ast_pbx_outgoing_exten() and
ast_pbx_outgoing_app() which could fail before setting the reason code.
This resulted in failures being reported as success.
ASTERISK-25266 #close
Reported by: Allen Ford
Change-Id: Idf16237b7e41b527d2c69c865829128686beeb3b
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
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
Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic
Edition after accepting the audio request but declining the video one.
Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c
Antoni Goldstein
Kevin Harwell
Sean Bright
Joshua C. Colp
Dömsödi Gergely
Rodrigo Ramírez Norambuena
George Joseph
Bryan Boatright
Alexei Gradinari
lvl
Corey Farrell
Richard Mudgett
Cao Minh Hiep
Ivan Poddubny
Robert Mordec
Kristian F. Høgh
Jenkins2
Sam Wierema
William McCall
Nic Colledge
Chris-Savinovich
Russell Bryant
Alexander Traud
Florian Floimair
Sungtae Kim
Michael Cargile
ghjm
Igor Goncharovskiy
Nathan Bruning
StefanEng86