AttendedTransfer queues up attended transfer to the given extension.
This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.
features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer
[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
same => n,Return()
[my_transfer]
include => default
;;;
This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:
features.conf
;;;
[featuremap]
atxfer => *7
[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer
[custom_atxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,AttendedTransfer(${dest})
same => n,Return()
[my_transfer]
include => default
;;;
Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b
BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.
This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.
features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer
[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
same => n,Return()
;;;
This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:
features.conf
;;;
[featuremap]
blindxfer =>
[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer
[custom_blindxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,BlindTransfer(${dest},default)
same => n,Return()
;;;
Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a
This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.
Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1
Various fixes for issues caught by gcc 9. Mostly snprintf
trying to copy to a buffer potentially too small.
ASTERISK-28412
Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e
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
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
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
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 1843b0e2b5.
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
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.
ao2_container_alloc is now restricted to modules only and is being
removed from Asterisk 17.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
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
* 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
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
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 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
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
As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.
ASTERISK-28046 #close
Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc
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
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
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
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
