Most of the configObjects and configOptions that are implemented with
ACO or Sorcery now have `<since>/<version>` elements added. There are
probably some that the script I used didn't catch. The version tags were
determined by the following...
* Do a git blame on the API call that created the object or option.
* From the commit hash, grab the summary line.
* Do a `git log --grep <summary>` to find the cherry-pick commits in all
branches that match.
* Do a `git patch-id` to ensure the commits are all related and didn't get
a false match on the summary.
* Do a `git tag --contains <commit>` to find the tags that contain each
commit.
* Weed out all tags not <major>.<minor>.0.
* Sort and discard any <major>.0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the API was last touched.
configObjects and configOptions elements implemented with the base
ast_config APIs were just not possible to find due to the non-deterministic
way they are accessed.
Also note that if the API call was on modified after it was added, the
version will be the one it was last modified in.
Final note: The configObject and configOption elements were introduced in
12.0.0 so options created before then may not have any XML documentation.
(cherry picked from commit 772221c82a)
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.
In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.
Fixes#546
Adds the end_marked_any option, which can be used
to kick a user from a conference if any marked user
leaves.
ASTERISK-30211 #close
Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5
If the CONFBRIDGE function is used to dynamically set
menu options, a memory leak occurs when a menu option
that has been set is overridden, since the menu entry
is not destroyed before being freed. This ensures that
it is.
Additionally, logic that duplicates the destroy function
is removed in lieu of the destroy function itself.
ASTERISK-28422 #close
Change-Id: I71cfb5c24e636984d41086d1333a416dc12ff995
Added the hear_own_join_sound option to the confbridge user profile to
control who hears the sound_join audio file. When set to 'yes' the user
entering the conference and the participants already in the conference
will hear the sound_join audio file. When set to 'no' the user entering
the conference will not hear the sound_join audio file, but the
participants already in the conference will hear the sound_join audio
file.
ASTERISK-29931
Added by Michael Cargile
Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2
A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.
ASTERISK-29440
Change-Id: I26642729d0345f178c7b8045506605c8402de54b
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
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
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
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
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
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
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
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
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
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
This change does a few things to improve packet loss and renegotiation:
1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.
2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.
3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.
4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.
5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.
ASTERISK-27143
Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
A previous commit added plumbing to bridge_softmix to allow for an SFU
experience with Asterisk. This commit adds an option to app_confbridge
that allows for a confbridge to actually make use of the SFU video mode.
SFU mode is implemented in a "set it and forget it" kind of way. That
is, when the bridge is created, if SFU mode is enabled, then the video
mode gets set to SFU and cannot be changed. Future improvements may
allow for a hybrid experience (e.g. forward multiple video streams,
specifically those of the most recent talkers), but for this addition,
no such capability is present.
Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020
menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function. In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case. aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made. Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.
* aco_process_config now sets info->internal->pending to NULL
after it unrefs it although this isn't strictly necessary in the
context of this fix.
* menu_template_handler now uses the "current" config and silently
ignores any attempt to be called as a result of someone uses the
"template" parameter in the conf file.
Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.
ASTERISK-25506 #close
Reported-by: Frederic LE FOLL
Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7
* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
Adds setting to confbridge.conf (binaural_active) that determines if binaural
synthesis can be available in bridge_softmix.
ASTERISK-26292
Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db
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
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
Confbridge dynamic profiles did not have a default profile unless you
explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a
template was not set prior to the bridge being created then some
options were left with no default values set. This patch makes it so
the default templates are set to the default bridge and user profiles.
ASTERISK-24749 #close
Reported by: philippebolduc
Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
When issuing a 'confbridge list XXXX' CLI command, the resulting output no
longer displays the menu associated with a ConfBridge participant.
The issue was caused by ASTERISK-22760. When that patch was done, it removed
the copying of the menu name associated with the user from the actual user
profile.
This patch fixes the issue by copying the menu name over to the user profile
when the menu hooks are applied to the user. Since that function now does a
little bit more than just apply the hooks, the name of the function has been
changed to cover the copying of the menu name over as well.
In addition, there is a disparity between the menu name length as it is stored
on the conf_menu structure and the confbridge_user structure; this patch makes
the lengths match so that a strcpy can be used.
Review: https://reviewboard.asterisk.org/r/4372/
ASTERISK-24723 #close
Reported by: Steve Pitts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.
New options are -
* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.
These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))
Review: https://reviewboard.asterisk.org/r/4023
ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
record_command-428838.patch uploaded by Gareth Palmer (License 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.
This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.
(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
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Merged revisions 407857 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 407858 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
Review: https://reviewboard.asterisk.org/r/2969/
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Merged revisions 402407 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402416 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also adds the ability to clear all profile items and makes behavior more
consistent with documentation as when choosing whether to use CONFBRIDGE
datastore profiles or the application arguments to the confbridge application.
(closes issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2971/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
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Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most hook callbacks did not need the bridge parameter. The pointer value
could become invalid if the channel is moved to another bridge while it is
executing.
* Fixed some issues in feature_attended_transfer() as a result.
* Reduce the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the bridge
channel hooks.
* Removed basic bridge requirement on feature_blind_transfer(). It does
not require the basic bridge like feature_attended_transfer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When this option was added, it was noted in CHANGES, but was missing
the XML documentation that this patch adds.
(closes issue ASTERISK-21780)
Patch-by: Brad Latus (snuffy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
