Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.
This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.
Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
conference (if the channel and conference use the same language)
ASTERISK-26289 #close
Reported by Mark Michelson
Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
In any scenario in which the callee is not connected to the caller, the
current code in app_dial will crash due to raising a Dial End Stasis
Message after the callee channel has been hung up. This patch corrects
the error by simply moving the explicit hangup of the callee (peer)
channel until after the dial end message.
ASTERISK-25691 #close
Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d
If the callee selects option '5' using the Dial application's privacy
(P) option, the DIALSTATUS is erroneously set to ANSWER. This option
reflects the callee sending the caller to VoiceMail one time; the call
is definitely *not* ANSWERed in such a scenario. With this patch, the
DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that
is set when the 'send to VoiceMail every time' option is set.
ASTERISK-25691
Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358
Previously, the buffer used for MP3 streamed from HTTP servers had a size of
1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1
minute. Only when the buffer is full does audio start to play.
For MP3 files streamed from a server, that is usually not a big deal as long as
the connection to the server is fast enough to supply that much data within a
second or two. For MP3 live streams however, it takes 1 minute to download 1
minute of audio, so without this change, app_mp3 wasn't really usable for MP3
live streams.
This commit changes the buffer size so that it covers 6 seconds of an MP3 file
streamed from a server and 0.5 seconds of an MP3 live stream. The latter is
identified by the use of a .m3u file extension.
app_mp3 so far only supported 8 kHz audio.
Now it always runs at the sample rate of the channel.
ASTERISK-26085 #close
Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0
When dialing channels it is possible that they may not ever
leave the not in use state (Local channels in particular) by
the time we cancel them. If this occurs but we know they were
dialed we explicitly remove them from the pending members
container so that subsequent call attempts occur.
ASTERISK-26299 #close
Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65
As described in issue ASTERISK-26282 the AEL parser creates macros with
extension '~~s~~'. app_macro searches only for extension 's' so the
created extension cannot be found. with this patch app_macro searches for
both extensions and performs the right extension.
ASTERISK-26282 #close
Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb
NOTE: This patch was submitted earlier and reverted because of a failing
test. The test has been patched so that it adjusts for the changes here,
so this is being resubmitted for review.
One feature that confbridge has is the ability to play sounds to all
participants in the conference. Prior to this commit, the algorithm for
this was as follows:
* Grab the playback lock
* Push the conference announcer channel into the bridge
* Play back the sound
* Pull the conference announcer channel from the bridge
* Release the playback lock
The issue here is that the act of adding the playback channel to the
bridge and removing it for each announcement is expensive. Amongst the
expenses:
* The announcer channel is imparted into the bridge, meaning a new
thread is spun up for each playback.
* When the announcer is added or removed from the bridge, it results
in the BRIDGEPEER channel variable being set on all channels in the
bridge. This requires keeping the bridge locked and locking each
individual channel in order to set it.
* There's also just the general overhead of adding the channel and
removing it from the bridge. The bridge potentially has to reconfigure
every single time
With this commit, the paradigm for playing back announcements has
shifted.
* The announcer channel is now added to the bridge when the conference
is allocated, and it is hung up when the conference is destroyed.
* A taskprocessor is used to queue playbacks onto the announcer channel.
This keeps the behavior from before where playbacks do not overlap.
* The announcer channel is no longer placed into the bridge as
departable. Since we are not constantly removing the channel from
the bridge, it is safe to add the channel using an independent thread
and simply hang the channel up when it is time for the conference to
be destroyed.
The use of the taskprocessor for playbacks opens up the interesting
possibility of having asynchronous announcements played. In this commit,
however, the behavior is still exactly the same as it previously was.
ASTERISK-26289
Reported by Mark Michelson
Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0
One feature that confbridge has is the ability to play sounds to all
participants in the conference. Prior to this commit, the algorithm for
this was as follows:
* Grab the playback lock
* Push the conference announcer channel into the bridge
* Play back the sound
* Pull the conference announcer channel from the bridge
* Release the playback lock
The issue here is that the act of adding the playback channel to the
bridge and removing it for each announcement is expensive. Amongst the
expenses:
* The announcer channel is imparted into the bridge, meaning a new
thread is spun up for each playback.
* When the announcer is added or removed from the bridge, it results
in the BRIDGEPEER channel variable being set on all channels in the
bridge. This requires keeping the bridge locked and locking each
individual channel in order to set it.
* There's also just the general overhead of adding the channel and
removing it from the bridge. The bridge potentially has to reconfigure
every single time
With this commit, the paradigm for playing back announcements has
shifted.
* The announcer channel is now added to the bridge when the conference
is allocated, and it is hung up when the conference is destroyed.
* A taskprocessor is used to queue playbacks onto the announcer channel.
This keeps the behavior from before where playbacks do not overlap.
* The announcer channel is no longer placed into the bridge as
departable. Since we are not constantly removing the channel from
the bridge, it is safe to add the channel using an independent thread
and simply hang the channel up when it is time for the conference to
be destroyed.
The use of the taskprocessor for playbacks opens up the interesting
possibility of having asynchronous announcements played. In this commit,
however, the behavior is still exactly the same as it previously was.
ASTERISK-26289
Reported by Mark Michelson
Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5
Some configuration directives were not initialized on reload, and hence
were not reset to default if they were removed from followme.conf.
ASTERISK-26288 #close
Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150
* Add some helpful <literal> and other embedded paragraph tags
* Document some of the lesser known channel variables set by Dial
* Add examples for some common Dial uses, along with some more
challenging but useful options
Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1
When a call forward attempt is made from a Queue member, the current
code will hang up the forwarding channel in an off-nominal condition
prior to raising the Stasis events informing the rest of Asterisk that
the call was forwarded. This will result in a slew of dreaded FRACKs,
most likely leading to a crash.
This patch modifies the code such that we don't hang up the forwarding
channel even in an off-nominal condition until we've safely raised the
Stasis messages.
ASTERISK-25797 #close
Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38
On heavy loaded system with IMAP or DB storage,
'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
It could happen when the IMAP or DB server dies or is unreachable.
It could happen on startup when there are many (thousands)
realtime endpoints configured with unsolicited mwi.
If the taskprocessor queue reaches the high water level
then the alert is triggered and pjsip stops processing new requests
until the queue reaches the low water level to clear the alert.
This patch adds 2 new 'general' configuration options
to tune taskprocessor alert levels:
'tps_queue_high' - Taskprocessor high water alert trigger level.
'tps_queue_low' - Taskprocessor low water clear alert level
ASTERISK-26229 #close
Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
code. This places all existing unit tests into a conditional block if
they weren't already.
ASTERISK-26211 #close
Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
This changes context includes from a linked list to a vector, makes
'struct ast_include' opaque to pbx.c.
Although ast_walk_context_includes is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_includes_count (AST_VECTOR_SIZE)
* ast_context_includes_get (AST_VECTOR_GET)
As with ast_walk_context_includes callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the includes, they have been converted to use the new functions.
const have been applied where possible to parameters for ast_include
functions.
Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60
It is possible for a not in use state change to occur multiple
times causing a queue member to be removed from the pending call
container prematurely.
The first not in use state change will remove the queue member
from the container. At this moment the member may be called and
placed in the pending container. After this another not in use
state change can be received which will remove it from the
container. Despite being called at this point the code will
incorrectly see that there are no pending calls to it.
This change only removes it from the pending container if the
state has actually changed.
ASTERISK-26133 #close
patches:
app_queue.diff submitted by Richard Miller (license 5685)
Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0
POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.
Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.
This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.
The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.
ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson
Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665
Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended
With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.
AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.
Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.
ASTERISK-25925 #close
Reported by Mark Michelson
Change-Id: I42cbec7730d84640a434d143a0d172a740995543
Fixed some bugs:
- create dirpath when save downloading message from IMAP storage.
- create IMAP folder if not exists when saving to IMAP storage
- check if file successfully opened before write to it
- some IMAP checks
- remove non-standard flag 'Unseen'
etc
Change to debug IMAP mm_status log instead of verbose.
Remove unused X-Asterisk-VM-Caller-channel message header
for security reason. The clients should not know name of peer/endpoint.
ASTERISK-26045 #close
Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b
Add the option 'enable_callee_prompt' to followme.conf. Enabled by
default. If disabled, a callee is not prompted to accept or reject
the forwarded call.
ASTERISK-26064 #close
Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
FollowMe with the option a records the name of the caller and plays it
to the callee. However it has failed to clean up that recorded file
as it tried to delete the file name without the '.sln' extension.
ASTERISK-26008 #close
Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip. This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.
ASTERISK-25989 #close
Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
When option 'o' was not set, ChanSpy created its audiohook with the flag
AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
direction only.
ASTERISK-25866 #close
Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0
Voicemail email addresses can be corrupt or voicemail
emails can end up being sent to the wrong email address if asterisk is
reading voicemail.conf during a reload and processing an email at the
same time. This patch always copies the struct that would otherwise only
be copied once.
ASTERISK-24463 #close
Reported by: John Campbell
Tested by: Etienne Lessard
Tested by: Andrew Nagy
Change-Id: I3a0643813116da84e2617291903d0d489b7425fb
ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC
and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when
queues grow too large or when read and write queues go out of sync.
Now these flags are set conditionally:
- AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set
- a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not
be set on the audiohook
ASTERISK-25866
Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd
When unloading the app_queue module the members in each queue are
destroyed and as part of this they are removed from the pending
members container. Unfortunately a crash would occur as the container
was destroyed before the members were removed.
This change tweaks ordering so the container destruction occurs
after the members are destroyed.
ASTERISK-16115
Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b
It was possible for a queue member that is a member of at least 2 or more
queues to receive mulitiple calls at the same time. This happened because
of a race between when a member was being rung and when the device state
notified the other queue(s) member object of the state change.
This patch makes it so when a queue member is being rung it gets added to
a global pool of queue members. If that same member is tried again, e.g.
from another queue, and it is found to already exist in the pending member
container then it will not ring that member.
ASTERISK-16115 #close
Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48
You cannot reference the passed in features struct after calling
ast_bridge_impart(). Even if the call fails.
Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21
This module is used as part of testsuite tests to confirm
stuff works. I'm accordingly marking it as core as it is
required by those tests.
Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88
The test_voicemail_notify_endl test checks the end-of-line
characters of an email message to confirm that they are consistent.
The test wrongfully assumed that reading from the email message
into a buffer will always result in more than 1 character being
read. This is incorrect. If only 1 character was read the test
would go outside of the buffer and access other memory causing
a crash.
The test now checks to ensure that 2 or more characters are read
in ensuring the test stays within the buffer.
ASTERISK-25874 #close
Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710
If try to move message to Cust1 (number 5)
the function 'save_to_folder' tries to create Greeting folder instead of Cust1.
This patch fixed it by setting GREETINGS_FOLDER = -1
ASTERISK-24927 #close
Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51
Sometimes uw-imap function 'mail_fetchbody' returns huge len
which then pass to uw-imap function 'rfc822_base64'.
uw-imap tries to allocate huge memory and abort() on fail.
This patch check the len.
If the len more than max size (128 Mbytes) log error.
This patch also set variables len, newlen to avoid uninizialezed len.
This patch also check pointer returned by rfc822_base64.
ASTERISK-25899 #close
Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca
This eliminates some casts that I made a note saying v10 and above
would no longer need them.
Better late than never :)
Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
When using app_echo via WebRTC with VP8 video the video would appear
only after a few minutes, because there would be nothing to request
a full reference frame.
This fixes the problem in both ways:
- echos any VIDUPDATE frames received on the channel
- sends one such frame when first video frame is to be forwarded
This makes the echo work with Firefox and Chrome WebRTC implementation.
ASTERISK-25867 #close
Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e
The configuration unsigned integer option handler sets flags for the
parser as if the option should be a signed integer (PARSE_INT32),
leading to errors on "out of range" values. Fix flags (PARSE_UINT32).
A fix to res_pjsip is also present which stops invalid flags from
being passed when registering sorcery object fields for qualify
status.
ASTERISK-25612 #close
Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e
This prevents pbx_core from hanging up the channel if the app isn't
registered.
ASTERISK-25846 #close
Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce
Channel masquerading had a conflict with autochannel locking.
When locking autochannel->channel, the channel is fetched from the
autochannel and then locked. During the fetch, the autochannel -- which
has no locks itself -- can be modified by someone who owns the channel
lock. That means that the value of autochan->channel cannot be trusted
until you hold the lock.
In practice, this caused problems with Local channels getting
masqueraded away while the ChanSpy attempted to get info from that
channel. The old channel which was about to get removed got locked, but
the new (replaced) channel got unlocked (no-op). Because the replaced
channel was now locked (and would never get unlocked), it couldn't get
removed from the channel list in a timely manner, and would now cause
deadlocks when iterating over the channel list.
This change checks the autochannel after locking the channel for changes
to the autochannel. If the channel had been changed, the lock is
reobtained on the new channel.
In theory it seems possible that after this fix, the lock attempt on the
old (wrong) channel can be on an already destroyed lock, maybe causing
a crash. But that hasn't been observed in the wild and is harder induce
than the current deadlock.
Thanks go to Filip Frank for suggesting a fix similar to this and
especially to IRC user hexanol for pointing out why this deadlock was
possible and testing this fix. And to Richard for catching my rookie
while loop mistake ;)
ASTERISK-25321 #close
Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def
Fix calculate of average time for talktime is wrong when is completed the
first call beacuse the time for talked would be that call.
ASTERISK-25800 #close
Change-Id: I94f79028935913cd9174b090b52bb300b91b9492
A user cannot set new bridge options after the conference is created by
the first user. Attempting to do so is documented as undefined behavior.
This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.
Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
* changes:
app_confbridge: Add ability to get the muted conference state.
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
app_confbridge: Make non-admin users join a muted conference muted.
* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
* Added Muted header to AMI ConfbridgeListRooms action response list
events to indicate the muted conference state.
* Added Muted column to CLI "confbridge list" output to indicate the muted
conference state and made the locked column a yes/no value instead of a
locked/unlocked value.
ASTERISK-20987
Reported by: hristo
Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1
Add time when started a the last pause for a queue member for
QueueMemberStatus ami event.
Also show accumulate time in seconds when started a pause for a queue
member to CLI command 'queue show'.
ASTERISK-16394 #close
Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c
When the Asterisk is restared is not preseved reason paused of members.
This patch fixed this cases, retain data on astdb and set when Asterisk
is started.
ASTERISK-25732 #close
Report by: Rodrigo Ramírez Norambuena
Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5
Member lastcall time is updated later than member status. There was chance to
check wrapuptime for available member with wrong (old) lastcall time.
New boolean flag "in_call" is set to true right before connecting call, and
reset to false after update of lastcall time. Members with "in_call" set to true
are treat as unavailable.
ASTERISK-19820 #close
Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500
The menuselect conflict between app_voicemail and res_mwi_external
makes it hard to package 1 version of Asterisk. There no actual
build dependencies between the 2 so moving this check to runtime
seems like a better solution.
The ast_vm_register and ast_vm_greeter_register functions in app.c
were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
is already a voicemail module registered. The modules' load_module
functions were then modified to return DECLINE instead of -1 to the
loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
the modules were incorrectly causing Asterisk to stop so this needed
to be cleaned up anyway.
Now you can build both and use modules.conf to decide which voicemail
implementation to load.
The default menuselect options still build app_voicemail and not
res_mwi_external but if both ARE built, res_mwi_external will load
first and become the voicemail provider unless modules.conf rules
prevent it. This is noted in CHANGES.
Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247
If a caller hangs up before dial is executed within an AGI then the AGI
has likely eaten all queued frames before executing the dial in DeadAGI
mode. With the caller hung up and no pending frames from the caller's
read queue, dial would not know that the call has hung up until a called
channel answers. It is rather annoying to whoever just answered the
non-existent call.
Dial should not continue execution in DeadAGI mode, hangup handlers, or
the h exten.
* Added a check early in dial to abort dialing if the caller has hungup.
ASTERISK-25307 #close
Reported by: David Cunningham
Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418
- The maximum_number_of_words was previously documented as being
the number of words that when exceeded, would result in the AMD
application returning that the audio represents a machine.
This was inconsistent with its actual functionality - it was
a number of words that when REACHED, would result in determination
as a machine.
This update corrects the functionality to match the previously
documented functionality. This is a backwards incompatible change
in configuration file, and has been added to UPGRADE.txt as a result.
The sample configuration file and application defaults have been updated
so that the default value is now 2, which reflects the same default
functionality as previous versions.
- Update documentation for silence_threshold, which previously implied
that it was measuring time, rather than noise averages in the sample.
- Update the comments in amd.conf.sample.
ASTERISK-25639 #close
Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093
If a call enters on a queue and the members on that queue are updated in
realtime (ex: using mysql inserting a new agent) the queue members are
never refreshed and the call will stay in the queue until other event occurs.
This happens only if this is the first call of the queue and there is no
agents servicing.
This patch prevent this issue, ensuring realtime members are updated if
there is one call in the queue and no available agents
ASTERISK-25442 #close
Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682
The default value was never set for audio_buffers, causing bad
audio quality. This ensures the default is always set.
ASTERISK-25569 #close
Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44
Add value of pause reason when is paused on CLI command "queue show"
ASTERISK-25581 #close
Report by: Rodrigo Ramírez Norambuena
Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864
To be able to barge into a call by dialling a prefix+extension that maps
to the extensions device.
Senario is that DECT headset users may be away from their desks and need
to transfer the call, the goal is that from any phone they dial a prefix
then their extension and are added to the bridge that they are in, from
there they can drop the headset call, as it's also on the handset,
and transfer the caller.
The dialplan would look like, where prefix=73, extension = 8512;
exten => _738512,1,BridgeAdd(SIP/cisco0001)
ASTERISK-25551 #close
Reported By: Alec Davis
Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540
Implemented support for the StatsD sample rate parameter,
which is a parameter for determining when to send computed
statistics to a client.
Valid sample rate values are:
Less than or equal to 0.0 will never be sent.
Between 0.0 and 1.0 will randomly be sent.
Greater than or equal to 1.0 will always be sent.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52
This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".
The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.
ASTERISK-25549 #close
Reported by Mark Michelson
Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
Added code to allow the StatsD dialplan application to
send data to the server specified in statsd.conf.
ASTERISK-25419
Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922
Added code to accept user input and validate it before
allowing it to be sent to the StatsD server.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3
In app_queue added value Paused Reason on QueueMemberStatus when a member
on queue is paused and the reason was set.
ASTERISK-25480 #close
Reporte by: Rodrigo Ramírez Norambuena
Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e
Wrote the skeleton framework for the Asterisk StatsD dialplan
application. This includes a load function, unload function, a
callback for execution, and XML documentation.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363
Mark Michelson
Matt Jordan
Michael Kuron
zuul
Joshua Colp
chrisderock
Tzafrir Cohen
Alexei Gradinari
Corey Farrell
Richard Mudgett
Timo Teräs
Örn Arnarson
Tzafrir Cohen
Jaco Kroon
Jean Aunis
Andrew Nagy
DarkS
Kevin Harwell
ibercom
Jacek Konieczny
Gianluca Merlo
Walter Doekes
Rodrigo Ramírez Norambuena
George Joseph
Martin Tomec
Dade Brandon
Carlos Oliva
Alec Davis
tcambron
Alexander Traud