-
ast_coredumper: check ast_debug_tools.conf permissions
ast_debug_tools.conf must be owned by root and not be
writable by other users or groups to be used by ast_coredumper or
by ast_logescalator or ast_loggrabber when run as root.
-
chan_websocket.conf.sample: Fix category name.
The category name in the chan_websocket.conf.sample file was
incorrect. It should be "global" instead of "general".
-
cli.c: Allow 'channel request hangup' to accept patterns.
The 'channel request hangup' CLI command now accepts
multiple channel names, POSIX Extended Regular Expressions, glob-like
patterns, or a combination of all of them. See the CLI command 'core
show help channel request hangup' for full details.
-
res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache populate command
The AMI command sorcery memory cache populate will now
return an error if there is an internal error performing the populate.
The CLI command will display an error in this case as well.
-
res_geolocation: Fix multiple issues with XML generation.
Geolocation: Two new optional profile parameters have been added.
pidf_element_id which sets the value of the id attribute on the top-level
PIDF-LO device, person or tuple elements.device_id which sets the content of the <deviceID> element.
Both parameters can include channel variables.
-
res_pjsip_messaging: Add support for following 3xx redirects
A new pjsip endpoint option follow_redirect_methods was added.
This option is a comma-delimited, case-insensitive list of SIP methods
for which SIP 3XX redirect responses are followed. An alembic upgrade
script has been added for adding this new option to the Asterisk
database.
-
taskprocessors: Improve logging and add new cli options
New CLI command has been added -
core show taskprocessor name
-
ccss: Add option to ccss.conf to globally disable it.
A new "enabled" parameter has been added to ccss.conf. It defaults
to "yes" to preserve backwards compatibility but CCSS is rarely used so
setting "enabled = no" in the "general" section can save some unneeded channel
locking operations and log message spam. Disabling ccss will also prevent
the func_callcompletion and chan_dahdi modules from loading.
-
Makefile: Add module-list-* targets.
Try "make module-list-deprecated" to see what modules
are on their way out the door.
-
app_mixmonitor: Add 's' (skip) option to delay recording.
This change introduces a new 's()' (skip) option to the MixMonitor
application. Example:
MixMonitor(${UNIQUEID}.wav,s(3))
This skips recording for the first 3 seconds before writing audio to the file.
Existing MixMonitor behavior remains unchanged when the 's' option is not used.
-
app_queue.c: Only announce to head caller if announce_to_first_user
When announce_to_first_user is false, no announcements are played to the head caller
-
res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
The STIR_SHAKEN_ATTESTATION dialplan function has been added
which will allow suppressing attestation on a call-by-call basis
regardless of the profile attached to the outgoing endpoint.
-
func_channel: Allow R/W of ADSI CPE capability setting.
CHANNEL(adsicpe) can now be read or written to change
the channels' ADSI CPE capability setting.
-
func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
Added a new option to HANGUPCAUSE to access additional
information about hangup reason. Reason headers from pjsip
could be read using 'tech_extended' cause type.
-
func_math: Add DIGIT_SUM function.
The DIGIT_SUM function can be used to return the digit sum of
a number.
-
app_sf: Add post-digit timer option to ReceiveSF.
The 't' option for ReceiveSF now allows for a timer since
the last digit received, in addition to the number-wide timeout.
-
app_dial: Allow fractional seconds for dial timeouts.
The answer and progress dial timeouts now have millisecond
precision, instead of having to be whole numbers.
-
chan_dahdi: Add DAHDI_CHANNEL function.
The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
-
app_queue.c: Add new global 'log_unpause_on_reason_change'
Add new global option 'log_unpause_on_reason_change' that
is default disabled. When enabled cause addition of UNPAUSE event on
every re-PAUSE with reason changed.
-
pbx_builtins: Allow custom tone for WaitExten.
The tone used while waiting for digits in WaitExten
can now be overridden by specifying an argument for the 'd'
option.
-
res_tonedetect: Add option for TONE_DETECT detection to auto stop.
The 'e' option for TONE_DETECT now allows detection to
be disabled automatically once the desired number of matches have
been fulfilled, which can help prevent race conditions in the
dialplan, since TONE_DETECT does not need to be disabled after
a hit.
-
sorcery: Prevent duplicate objects and ensure missing objects are created on update
Users relying on Sorcery multiple writable backends configurations
(e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
in sorcery.conf to ensure missing objects are recreated after temporary backend
failures. Default behavior remains unchanged unless explicitly enabled.
-
chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
A new WebSocket channel driver option v has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run core show application Dial from the Asterisk CLI
to see how to use it.
-
app_chanspy: Add option to not automatically answer channel.
ChanSpy and ExtenSpy can now be configured to not
automatically answer the channel by using the 'N' option.
-
cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
Enabling the tracking of the
STREAM_BEGIN and the STREAM_END event
types in cel.conf will log media files and
music on hold played to each channel.
The STREAM_BEGIN event's extra field will
contain a JSON with the file details (path,
format and language), or the class name, in
case of music on hold is played. The DTMF
event's extra field will contain a JSON with
the digit and the duration in milliseconds.
-
res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
Options are now available in the menuselect "Resource Modules"
category that allow you to enable the AES_192, AES_256 and AES_GCM
cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
them but modern versions do. Previously, the only way to enable them was
to set the CFLAGS environment variable when running ./configure.
The default setting is to disable them preserving existing behavior.
-
cdr: add CANCEL dispostion in CDR
A new CDR option "canceldispositionenabled" has been added
that when set to true, the NO ANSWER disposition will be split into
two dispositions: CANCEL and NO ANSWER. The default value is 'no'
-
func_curl: Allow auth methods to be set.
The httpauth field in CURLOPT now allows the authentication
methods to be set.
-
Media over Websocket Channel Driver
A new channel driver "chan_websocket" is now available. It can
exchange media over both inbound and outbound websockets and will both frame
and re-time the media it receives.
See http://s.asterisk.net/mow for more information.
The ARI channels/externalMedia API now includes support for the
-
res_stir_shaken.so: Handle X5U certificate chains.
The STIR/SHAKEN verification process will now load a full
certificate chain retrieved via the X5U URL instead of loading only
the end user cert.
-
res_stir_shaken: Add "ignore_sip_date_header" config option.
A new STIR/SHAKEN verification option "ignore_sip_date_header" has
been added that when set to true, will cause the verification process to
not consider a missing or invalid SIP "Date" header to be a failure. This
will make the IAT the sole "truth" for Date in the verification process.
The option can be set in the "verification" and "profile" sections of
stir_shaken.conf.
Also fixed a bug in the port match logic.
Resolves: #1251
Resolves: #1271
-
app_record: Add RECORDING_INFO function.
The RECORDING_INFO function can now be used
to retrieve the duration of a recording.
-
app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range
This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises
only for members whose current penalty is within the [min_penalty, max_penalty] range.
Members with lower or higher penalties are unaffected.
This behavior is backward-compatible with existing queue rule configurations.
-
res_odbc: cache_size option to limit the cached connections.
New cache_size option for res_odbc to on a per class basis limit the
number of cached connections. Please reference the sample configuration
for details.
-
res_odbc: cache_type option for res_odbc.
When using res_odbc it should be noted that back-end
connections to the underlying database can now be configured to re-use
the cached connections in a round-robin manner rather than repeatedly
re-using the same connection. This helps to keep connections alive, and
to purge dead connections from the system, thus more dynamically
adjusting to actual load. The downside is that one could keep too many
connections active for a longer time resulting in resource also begin
consumed on the database side.
-
ARI Outbound Websockets
Asterisk can now establish websocket sessions to your ARI applications
as well as accepting websocket sessions from them.
Full details: http://s.asterisk.net/ari-outbound-ws
-
res_websocket_client: Create common utilities for websocket clients.
A new module "res_websocket_client" and config file
"websocket_client.conf" have been added to support several upcoming new
capabilities that need common websocket client configuration.
-
asterisk.c: Add option to restrict shell access from remote consoles.
A new asterisk.conf option 'disable_remote_console_shell' has
been added that, when set, will prevent remote consoles from executing
shell commands using the '!' prefix.
Resolves: #GHSA-c7p6-7mvq-8jq2
-
sig_analog: Add Call Waiting Deluxe support.
Call Waiting Deluxe can now be enabled for FXS channels
by enabling its corresponding option.
-
stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
A Dial timeout on POST /channels/{channelId}/dial will now result in a
CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously
no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
-
contrib: Add systemd service and timer files for malloc trim.
Service and timer files for systemd have been added to the
contrib/systemd/ directory. If you are experiencing memory issues,
install these files to have "malloc trim" periodically run on the
system.
-
Add log-caller-id-name option to log Caller ID Name in queue log
This patch adds a global configuration option, log-caller-id-name, to queues.conf
to control whether the Caller ID name is logged as parameter 4 when a call enters a queue.
When log-caller-id-name=yes, the Caller ID name is included in the queue log,
Any '|' characters in the caller ID name will be replaced with '_'.
(provided it’s allowed by the existing log_restricted_caller_id rules).
When log-caller-id-name=no (the default), the Caller ID name is omitted.
-
asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
In cli.conf, you can now define startup commands that run before
core initialization and before module initialization.
-
audiosocket: added support for DTMF frames
The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).
-
ari/pjsip: Make it possible to control transfers through ARI
Call transfers on the PJSIP channel can now be controlled by
ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
dialplan function.
-
sig_analog: Add Last Number Redial feature.
Users can now redial the last number
called if the lastnumredial setting is set to yes.
Resolves: #437
-
Add SHA-256 and SHA-512-256 as authentication digest algorithms
The SHA-256 and SHA-512-256 algorithms are now available
for authentication as both a UAS and a UAC.
-
Upgrade bundled pjproject to 2.15.1 Resolves: #1016
Bundled pjproject has been upgraded to 2.15.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
res_pjsip: Add new AOR option "qualify_2xx_only"
The pjsip.conf AOR section now has a "qualify_2xx_only"
option that can be set so that only 2XX responses to OPTIONS requests
used to qualify a contact will mark the contact as available.
-
app_queue: allow dynamically adding a queue member in paused state.
use the p option of AddQueueMember() for paused member state.
Optionally, use the r(reason) option to specify a custom reason for the pause.
-
manager.c: Add Processed Call Count to CoreStatus output
The current processed call count is now returned as CoreProcessedCalls from the
CoreStatus AMI Action.
-
func_curl.c: Add additional CURL options for SSL requests
The following new configuration options are now available
in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost'
(CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath'
(CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype'
(CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
(CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the
libcurl documentation for more details.
-
res_stir_shaken: Allow sending Identity headers for unknown TNs
You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.
-
manager.c: Restrict ListCategories to the configuration directory.
The ListCategories AMI action now restricts files to the
configured configuration directory.
-
res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
app_mixmonitor: Add 'D' option for dual-channel audio.
The MixMonitor application now has a new 'D' option which
interleaves the recorded audio in the output frames. This allows for
stereo recording output with one channel being the transmitted audio and
the other being the received audio. The 't' and 't' options are
compatible with this.
-
manager.c: Restrict ModuleLoad to the configured modules directory.
The ModuleLoad AMI action now restricts modules to the
configured modules directory.
-
manager: Enhance event filtering for performance
You can now perform more granular filtering on events
in manager.conf using expressions like
eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/
This is much more efficient than
eventfilter = Event: Newchannel.*Channel: PJSIP/
Full syntax guide is in configs/samples/manager.conf.sample.
-
db.c: Remove limit on family/key length
The ast_db_*() APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the DB* dialplan applications, dialplan functions,
manager actions and databse CLI commands. Since the
media_cache also uses the ast_db_*() APIs, you can now store
resources with URIs longer than 253 bytes.
-
res_pjsip_notify: add dialplan application
A new dialplan application PJSIPNotify is now available
which can send SIP NOTIFY requests from the dialplan.
The pjsip send notify CLI command has also been enhanced to allow
sending NOTIFY messages to a specific channel. Syntax:
pjsip send notify channel
-
channel: Add multi-tenant identifier.
tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
-
feat: ARI "ChannelToneDetected" event
Setting the TONE_DETECT dialplan function on a channel
in ARI will now cause a ChannelToneDetected ARI event to be raised
when the specified tone is detected.
-
res_pjsip_config_wizard.c: Refactor load process
The res_pjsip_config_wizard.so module can now be reloaded.
-
app_voicemail_odbc: Allow audio to be kept on disk
This commit adds a new voicemail.conf option
'odbc_audio_on_disk' which when set causes the ODBC variant of
app_voicemail_odbc to leave the message and greeting audio files
on disk and only store the message metadata in the database.
Much more information can be found in the voicemail.conf.sample
file.
-
app_queue: Add option to not log Restricted Caller ID to queue_log
Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
will be stored in the queue log.
If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
-
pbx.c: expand fields width of "core show hints"
The fields width of "core show hints" were increased.
The width of "extension" field to 30 characters and
the width of the "device state id" field to 60 characters.
-
rtp_engine: add support for multirate RFC2833 digits
No change in configuration is required in order to enable this
feature. Endpoints configured to use RFC2833 will automatically have this
enabled. If the endpoint does not support this, it should not include it in
the SDP offer/response.
Resolves: #699
-
res_pjsip_logger: Preserve logging state on reloads.
Issuing "pjsip reload" will no longer disable
logging if it was previously enabled from the CLI.
-
loader.c: Allow dependent modules to be unloaded recursively.
In certain circumstances, modules with dependency relations
can have their dependents automatically recursively unloaded and loaded
again using the "module refresh" CLI command or the ModuleLoad AMI command.
-
tcptls/iostream: Add support for setting SNI on client TLS connections
Secure websocket client connections now send SNI in
the TLS client hello.
-
res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
this new feature let users match endpoints based on the
indound SIP requests' URI. To do so, add 'request_uri' to the
endpoint's 'identify_by' option. The 'match_request_uri' option of
the identify can be an exact match for the entire request uri, or a
regular expression (between slashes). It's quite similar to the
header identifer.
Fixes: #599
-
res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.
-
manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication:
[addautoanswer]
exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
exten => _s,n,Return()
-
manager.c: Add CLI command to kick AMI sessions.
The "manager kick session" CLI command now
allows kicking a specified AMI session.
-
chan_dahdi: Allow specifying waitfordialtone per call.
"waitfordialtone" may now be specified for DAHDI
trunk channels on a per-call basis using the CHANNEL function.
-
Upgrade bundled pjproject to 2.14.1
Bundled pjproject has been upgraded to 2.14.1. For more
information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
app_dial: Add dial time for progress/ringing.
The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
-
app_voicemail: Allow preventing mark messages as urgent.
The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
-
Stir/Shaken Refactor
Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
-
Upgrade bundled pjproject to 2.14.
Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
-
app_speech_utils.c: Allow partial speech results.
The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
-
res_pjsip_outbound_registration.c: Add User-Agent header override
PJSIP outbound registrations now support a per-registration
User-Agent header
-
app_chanspy: Add 'D' option for dual-channel audio
The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
-
app_voicemail_odbc: remove macrocontext from voicemail_messages table
The fix requires removing the macrocontext column from the
voicemail_messages table in the voicemail database via alembic upgrade.
-
chan_dahdi: Allow MWI to be manually toggled on channels.
The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440
-
app_dial: Add option "j" to preserve initial stream topology of caller
The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.
-
logger: Add channel-based filtering.
The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
-
chan_pjsip: Add PJSIPHangup dialplan app and manager action
A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
-
app_voicemail: Add AMI event for mailbox PIN changes.
The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
Resolves: #398
-
res_speech: allow speech to translate input channel
res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
-
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
-
res_stasis: signal when new command is queued
Call setup times should be significantly improved
when using ARI.
-
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
-
file.c: Add ability to search custom dir for sounds
A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/ directory.
-
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
-
chan_rtp: Implement RTP glue for UnicastRTP channels
The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
-
variables: Add additional variable dialplan functions.
Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.
-
sig_analog: Add Called Subscriber Held capability.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
-
res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
-
core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint /endpoints/refer for referring
an endpoint to some URI or endpoint.
-
chan_dahdi: Allow autoreoriginating after hangup.
The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
-
sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
-
res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
-
app_queue: Add support for applying caller priority change immediately.
The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
-
Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
The following manager actions have been added
VoicemailBoxSummary - Generate message list for a given mailbox
VoicemailRemove - Remove a message from a mailbox folder
VoicemailMove - Move a message from one folder to another within a mailbox
VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.
-
app_voicemail: add CLI commands for message manipulation
The following CLI commands have been added to app_voicemail
voicemail show mailbox
Show contents of mailbox @
voicemail remove <from_folder>
Remove message from <from_folder> in mailbox @
voicemail move <from_folder> <to_folder>
Move message in mailbox & from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <to_mailbox> <to_context> <to_folder>
Forward message in mailbox @ <from_folder> to
mailbox @ <to_folder>
-
sig_analog: Allow immediate fake ring to be suppressed.
The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
-
AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.
-
res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
-
AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.
-
sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
-
res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.
-
app_queue: Preserve reason for realtime queues
Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
-
cel: add local optimization begin event (#54)
The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
-
chan_dahdi: Add dialmode option for FXS lines.
A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
-
http.c: Change httpstatus to default disabled and sanitize output.
To prevent possible security issues, the /httpstatus page
served by the internal web server is now disabled by default. To explicitly
enable it, set enable_status=yes in http.conf.
-
res_geolocation: Fix multiple issues with XML generation.
Geolocation: In order to correct bugs in both code and
documentation, the following changes to the parameters for GML geolocation
locations are now in effect:
- The documented but unimplemented
crs (coordinate reference system) element
has been added to the location_info parameter that indicates whether the 2d
or 3d reference system is to be used. If the crs isn't valid for the shape
specified, an error will be generated. The default depends on the shape
specified.
- The Circle, Ellipse and ArcBand shapes MUST use a
2d crs. If crs isn't
specified, it will default to 2d for these shapes.
The Sphere, Ellipsoid and Prism shapes MUST use a 3d crs. If crs isn't
specified, it will default to 3d for these shapes.
The Point and Polygon shapes may use either crs. The default crs is 2d
however so if 3d positions are used, the crs must be explicitly set to 3d.
- The
geoloc show gml_shape_defs CLI command has been updated to show which
coordinate reference systems are valid for each shape.
- The
pos3d element has been removed in favor of allowing the pos element
to include altitude if the crs is 3d. The number of values in the pos
element MUST be 2 if the crs is 2d and 3 if the crs is 3d. An error
will be generated for any other combination.
- The angle unit-of-measure for shapes that use angles should now be included
in the respective parameter. The default is degrees. There were some
inconsistent references to orientation_uom in some documentation but that
parameter never worked and is now removed. See examples below.
Examples...
location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
location_info = shape="Point", pos="39.0 -105.0"
location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
pidf_element_id = ${CHANNEL(name)}-${EXTEN}
device_id = mac:001122334455
Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-
app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
In an effort to reduce log spam, two normal progress
"pickup attempted" log messages from app_directed_pickup have been changed
from NOTICE to VERBOSE(3). This puts them on par with other normal
dialplan progress messages.
-
app_queue.c: Fix error in Queue parameter documentation.
As part of Asterisk 21, macros were removed from Asterisk.
This resulted in argument order changing for the Queue dialplan
application since the macro argument was removed. Upgrade notice was
missed when this was done, so this upgrade note has been added to
provide a record of such and a notice to users who may have not upgraded
yet.
-
res_audiosocket: add message types for all slin sample rates
New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.
-
taskpool: Add taskpool API, switch Stasis to using it.
The threadpool_* options in stasis.conf have now been deprecated
though they continue to be read and used. They have been replaced with taskpool
options that give greater control over the underlying taskpool used for stasis.
-
safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
The safe_asterisk script now checks that, if it was run by the
root user, the /etc/asterisk/startup.d directory and all the files it contains
are owned by root. If the checks fail, safe_asterisk will exit with an error
and Asterisk will not be started. Additionally, the default logging
destination is now stderr instead of tty "9" which probably won't exist
in modern systems.
-
jansson: Upgrade version to jansson 2.14.1
jansson has been upgraded to 2.14.1. For more
information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.14.1
-
Alternate Channel Storage Backends
With this release, you can now select an alternate channel
storage backend based on C++ Maps. Using the new backend may increase
performance and reduce the chances of deadlocks on heavily loaded systems.
For more information, see http://s.asterisk.net/dc679ec3
-
ARI: REST over Websocket
This commit adds the ability to make ARI REST requests over the same
websocket used to receive events.
See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
alembic: Database updates required.
Two commits in this release...
'Add SHA-256 and SHA-512-256 as authentication digest algorithms'
'res_pjsip: Add new AOR option "qualify_2xx_only"'
...have modified alembic scripts for the following database tables: ps_aors,
ps_contacts, ps_auths, ps_globals. If you don't use the scripts to update
your database, reads from those tables will succeeed but inserts into the
ps_contacts table by res_pjsip_registrar will fail.
-
channel: Add multi-tenant identifier.
A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
-
app_queue: Add option to not log Restricted Caller ID to queue_log
Add a new column to the queues table:
queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
to control whether the Restricted Caller ID will be stored in the queue log.
-
pbx_variables.c: Prevent SEGV due to stack overflow.
The maximum amount of dialplan recursion
using variable substitution (such as by using EVAL_EXTEN)
is capped at 15.
-
Stir/Shaken Refactor
The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
-
app_voicemail_odbc: remove macrocontext from voicemail_messages table
The fix requires that the voicemail database be upgraded via
alembic. Upgrading to the latest voicemail database via alembic will
remove the macrocontext column from the voicemail_messages table.
-
app.c: Allow ampersands in playback lists to be escaped.
Ampersands in URLs passed to the Playback(),
Background(), SpeechBackground(), Read(), Authenticate(), or
Queue() applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the CONFBRIDGE dialplan function, or configuring various
features in confbridge.conf and queues.conf.
-
pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.
-
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.
-
users.conf: Deprecate users.conf configuration.
The users.conf config is now deprecated
and will be removed in a future version of Asterisk.
-
app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
-
app_sla: Migrate SLA applications out of app_meetme.
The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.
-
utils.h: Deprecate ast_gethostbyname(). (#79)
ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use ast_sockaddr_resolve() and
ast_sockaddr_resolve_first_af().
-
cel: add local optimization begin event (#54)
The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
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