Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.
ASTERISK-29336
Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35
This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.
ASTERISK-29335
Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90
Queue members using dialplan hints as a state interface must handle
INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE.
ASTERISK-28369
Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close
Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
launch_monitor_thread is responsible for creating and initializing
the mixmonitor, and dependent data structures. There was one off
nominal path after the datastore gets created that triggers when
the channel being monitored is hung up prior to monitor starting
itself.
If this happened the monitor thread would not "launch", and the
mixmonitor object and associated objects are freed, including the
underlying datastore data object. However, the datastore itself was
not removed from the channel, so when the channel eventually gets
destroyed it tries to access the previously freed datastore data
and crashes.
This patch removes and frees datastore object itself from the channel
before freeing the mixmonitor object thus ensuring the channel does
not call it when destroyed.
ASTERISK-28947 #close
Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8
The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.
https://wiki.asterisk.org/wiki/x/Xc5uAg
However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.
For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.
ASTERISK-28883 #close
Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f
When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.
ASTERISK-29118 #close
Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb
Operations that update queues when shared_lastcall is set lock the
queue in question, then have to lock the queues container to find the
other queues with the same member. On the other hand, __queues_show
(which is called by both the CLI and AMI) does the reverse. It locks
the queues container, then iterates over the queues locking each in
turn to display them. This creates a deadlock.
* Moved queue print logic from __queues_show to a separate function
that can be called for a single queue.
* Updated __queues_show so it doesn't need to lock or traverse
the queues container to show a single queue.
* Updated __queues_show to snap a copy of the queues container and iterate
over that instead of locking the queues container and iterating over
it while locked. This prevents us from having to hold both the
container lock and the queue locks at the same time. This also
allows us to sort the queue entries.
ASTERISK-29155
Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41
app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge. To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second. The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".
Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a
This fixes a bug introduced mistakenly in ASTERISK-25665:
If leave-empty is enabled, a call may sometimes be removed from
a queue without recording it as abandoned.
This causes Asterisk to not generate an abandon event for that
call, and for the queue abandoned counter to be incorrect.
ASTERISK-29043 #close
Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7
This fixes the reseting members lastpause problem when realtime members is being used,
the function rt_handle_member_record was forcing the reset members lastpause because it
does not exist in realtime
ASTERISK-29034 #close
Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5
Rather than putting messages into INBOX and then moving them to Urgent
later, put them directly in to the Urgent folder. This prevents
mailcmd from being skipped.
ASTERISK-27273 #close
Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5
The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
at the same level as the scope level. This allows the same
messages to be printed to the debug log when AST_DEVMODE
isn't enabled.
Also added a few variants of the SCOPE_EXIT macros that will
also call ast_log instead of ast_debug to make it easier to
use scope tracing and still print error messages.
Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21
Allow passing a topology from the called channel back to the
calling channel.
* Added a new function ast_queue_answer() that accepts a stream
topology and queues an ANSWER CONTROL frame with it as the
data. This allows the called channel to indicate its resolved
topology.
* Added a new virtual function to the channel tech structure
answer_with_stream_topology() that allows the calling channel
to receive the called channel's topology. Added
ast_raw_answer_with_stream_topology() that invokes that virtual
function.
* Modified app_dial.c and features.c to grab the topology from the
ANSWER frame queued by the answering channel and send it to
the calling channel with ast_raw_answer_with_stream_topology().
* Modified frame.c to automatically cleanup the reference
to the topology on ANSWER frames.
Added a few debugging messages to stream.c.
Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c
If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.
(Actually, commit ed615afb7e already
partially fixed it for shared_lastcall.)
ASTERISK-28951
Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.
No functional changes were made with this commit.
Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.
This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.
ASTERISK-28954
Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56
Because ring_entry() is not called, outgoing->chan is not touched here
either.
ASTERISK-28950
ASTERISK-28644
Change-Id: I564613715dfaf45af868251eb75a451f512af90f
Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.
This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.
In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.
ASTERISK-28952
Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3
When send_events is enabled for a user, we were leaking a reference
to the bridge channel in confbridge_manager.c:send_message(). This
also caused the bridge snapshot to not be destroyed.
Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97
This patch fixes a few compile warnings/errors that now occur when using gcc
10+.
Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.
Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9
Ensure that output buffers for the osp_convert_inout
function have sufficient space for additional data
such as brackets and ports.
ASTERISK-28804
Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b
Add a new "masquarade" channel event, and use it in app_queue to track unique id's.
Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210
ASTERISK-28829 #close
ASTERISK-25844 #close
Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
The gcc 10 -Wrestrict option was causing "overlap" errors when
snprintf was copying one char[256] structure member to another
char[256] member in the same structure.
Using ast_alloca instead of declaring the structure inline
solves the issue.
Here's a link to the "enhancement":
https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html
We may follow up with a gcc bug report.
Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534
Since Asterisk 14, app_fax did not compile at all because Asterisk
requires that not malloc but ast_malloc is used everywhere. However,
the system headers of SpanDSP use malloc. Because we cannot (and do
not need to) change system headers, let us ignore this.
ASTERISK-28848
Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546
When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".
ASTERISK-28841
Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13
Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.
This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.
ASTERISK-28780
Change-Id: I8044c06daa06f0f16607788c596f55623be26f58
The optional synchronization behavior created in
64906c4c9b is now the default for
MixMonitor.
* Add a new flag 'n' that allows for this behavior to be turned off
* Add a notice when the 'S' option is used indicating that it is no
longer necessary
Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4
When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.
* If the user wants to write to a wav49 file, make sure that it is
reflected properly in MIXMONITOR_FILENAME.
* Add a note to the documentation describing this behavior.
* Add a note in main/file.c indicating that app_mixmonitor needs to be
changed if the logic in build_filename was changed.
ASTERISK-24798 #close
Reported by: xrobau
Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c
* The MailboxExists dialplan application was deprecated on 2006-09-26
in Asterisk 1.6.0 (commit ec83b11183)
* The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in
Asterisk 11.0.0 (commit fd64bb66f9)
Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f
In af90afd90c, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.
Whitespace only, no functional change.
ASTERISK~23324
Reported by: Kevin McCoy
Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87
ast_store_realtime() is not NULL tolerant, so we need to initialize
the field values we pass to it to the empty string to avoid a crash.
ASTERISK-23739 #close
Reported by: Stas Kobzar
Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c
If voicemail.conf exists but is empty, the config parsing process will
default a number of global variables to non-zero values. On the other
hand, if voicemail.conf is missing (arguably semantically equivalent
to an empty file), this process is skipped and the globals are
defaulted to 0.
Set the globals to the same values they would be set to if a
configuration were present. This allows voicemail configuration to be
done completely by Realtime without the need to create an empty
voicemail.conf file.
ASTERISK-27622 #close
Reported by: Jim Van Meggelen
Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409
This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.
A description of the protocol can be found on the above referenced
GitHub page. A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
ASTERISK-28484 #close
Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
The QueueMemberPause AMI event includes two fields that return the
reason a member was paused.
* In release branches, deprecate Reason in favor of PausedReason.
* In master, remove the Reason field entirely.
ASTERISK-28349 #close
Reported by: Niksa Baldun
Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296
Additionally alter the warning to mention that it was "beep" which could
not be streamed to give admins a better clue about what the warning
means.
ASTERISK-28682
Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e
This patch fixes some wrongly formatted documentation for the AgentLogin
application. A couple of "see also" links should contain only the function
name, and no parameters.
Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661
Dialplan has to be careful about passing an empty device list or empty
positions in the list. As a result, dialplan has to check for these
conditions before using ChanIsAvail. Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.
* Made tolerate empty positions in the device list.
* Simplified the code and eliminated some unnecessary indention.
ASTERISK-28638
Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3
* Made BridgeAdd not hangup the call if there is a problem.
* Reduced message level from warning to verbose for normal exception
cases.
* Added a loop safety check to BridgeAdd.
* Made BridgeAdd set BRIDGERESULT with the status when dialplan is
resumed.
Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Dial. Simplify dialplan by making Dial
handle these conditions gracefully.
* Made tolerate empty positions in the dialed device list.
* Reduced some message log levels from notice to verbose.
ASTERISK-28638
Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Page. Simplify dialplan by making Page
handle these conditions gracefully.
* Made tolerate empty positions in the paged device list.
* Reduced some warnings associated with the 's' option to verbose
messages. The warning level for those messages really serves no purpose
as that is why the 's' option exists.
ASTERISK-28638
Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3
Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.
ASTERISK-28636
Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.
* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began
ASTERISK-28608
Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
The following modules needed tweaks for API changes.
addons/cdr_mysql.c
addons/chan_ooh323.c
apps/app_meetme.c
ASTERISK-28604
Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d
If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.
ASTERISK-22192
Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd
This reverts commit fd2e8d0da7.
Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.
ASTERISK-28567 #close
Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41
ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.
This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().
ASTERISK-28527
Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.
ASTERISK-28505 #close
Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c
You now select voicemail backends like normal dialplan applications, so
there is no longer a need for their own menuselect category.
Reported by snuff-work in #asterisk-dev
Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:
unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);
would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.
ASTERISK-28480
Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1
app_voicemail utilized the stasis cache when polling mailboxes for MWI. This
caused a memory leak (items were not being appropriately removed from the
cache), and subsequent slowdown in system processing. This patch removes the
stasis cache dependency, thus alleviating the memory leak. It does this by
utilizing the new MWI API that better manages state lifetime.
ASTERISK-28443
ASTERISK-27121
Change-Id: Ie89fedaca81ea1fd03d150d9d3a1ef3d53740e46
Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.
Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7
Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual
wait time instead of `framelength` is enough to fix timeout / TOOLONG issue.
ASTERISK-28419 #close
Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb
When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.
Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1
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
When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.
This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.
ASTERISK-28401
Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c
There a long history here:
In commit dd1e62c095 has introduce by default shared_lastcall = true by
default but this now only happen is there not [general] directive in
queues.conf
After that, the commit 4b50e3f1ee fix the
sample file.
We'll need to keep the same setting if there a general or not section in
configuration file since the shared_lastcall is by a long time in
sample files as default value to 'no'.
Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c
Joshua C. Colp
Sean Bright
Ivan Poddubnyi
Sebastien Duthil
Dan Cropp
Kevin Harwell
George Joseph
Alexander Traud
Kfir Itzhak
Evandro César Arruda
Walter Doekes
Kevin Harwell
Nathan Bruning
Jaco Kroon
Friendly Automation
Seán C McCord
Corey Farrell
Richard Mudgett
Frederic LE FOLL
Michael Cargile
lvl
cmaj
Alexei Gradinari
Chris-Savinovich
Nasir Iqbal
Joshua Colp
agupta
Rodrigo Ramírez Norambuena