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pkgsrc changes:
- adapt to various upstream changes
- update for newer version of pjproject
- add unconditional depeendency on SDL
- remove pktccops and mgcp option (has to do with supporting cable headends)
- remove various 64-bit time_t fixes as upstream is finally doing these
[asterisk-announce] asterisk release 18.21.0
The Asterisk Development Team would like to announce
the release of asterisk-18.21.0.
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release asterisk-18.21.0
========================================
Summary:
----------------------------------------
- logger: Fix linking regression.
- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
User Notes:
----------------------------------------
- ### 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/<lang> 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.
- ### app_queue: periodic announcement configurable start time.
Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue. The default behavior if this config option is
not set remains unchanged.
Signed-off-by: Jaco Kroon <jaco at uls.co.za>
- ### 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.
Upgrade Notes:
----------------------------------------
- ### 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.
[asterisk-announce] asterisk release 18.20.2
The Asterisk Development Team would like to announce
the release of asterisk-18.20.2.
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release asterisk-18.20.2
========================================
Summary:
----------------------------------------
- res_rtp_asterisk: Fix regression issues with DTLS client check
[asterisk-announce] asterisk release 18.20.1
The Asterisk Development Team would like to announce security release
Asterisk 18.20.1.
The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/s
ecurity/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github
.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asteris
k/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com
/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
Change Log for Release asterisk-18.20.1
========================================
Summary:
----------------------------------------
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
[asterisk-announce] asterisk release 18.20.0
The Asterisk Development Team would like to announce
the release of asterisk-18.20.0.
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release asterisk-18.20.0
========================================
Summary:
----------------------------------------
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- 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.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using
`rtp->themssrc_valid` into the scope of the rtp_i nstance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
User Notes:
----------------------------------------
- ### 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 Vo icemailBoxSummarry, 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 <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder>
<messageid> <to_mailbox> <to_context> <to_ folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <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.
[asterisk-announce] Asterisk Release 18.19.0
The Asterisk Development Team would like to announce
the release of Asterisk 18.19.0.
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release 18.19.0
========================================
Summary:
----------------------------------------
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
User Notes:
----------------------------------------
- ### 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.
Upgrade Notes:
----------------------------------------
- ### 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.
Closed Issues:
----------------------------------------
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect
.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.
[asterisk-announce] Asterisk Release 18.18.1
The Asterisk Development Team would like to announce security release
Asterisk 18.18.1.
The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
Change Log for Release 18.18.1
========================================
Summary:
----------------------------------------
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
User Notes:
----------------------------------------
- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs
- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
[asterisk-announce] Asterisk Release 18.18.0
The Asterisk Development Team would like to announce
the release of Asterisk 18.18.0.
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release 18.18.0
========================================
Summary:
----------------------------------------
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
User Notes:
----------------------------------------
- ### cel: add local optimization begin event
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. (#36)
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.
- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs
- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
Upgrade Notes:
----------------------------------------
- ### cel: add local optimization begin event
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.
Closed Issues:
----------------------------------------
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
### For more details, see:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md
[asterisk-announce] Asterisk 18.17.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.17.1.
The release of Asterisk 18.17.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30469 - res_pjsip_pubsub: Regression for
subscription shutdowns
(Reported by N A)
* ASTERISK-30472 - pbx_ael: Literal usage for variables broken
(Reported by isrl)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
Thank you for your continued support of Asterisk!
[asterisk-announce] Asterisk 18.17.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.17.0.
The release of Asterisk 18.17.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-29810 - app_signal: Add channel signaling
applications
(Reported by N A)
* ASTERISK-30262 - res_pjsip_session: Allow a context to be
specified for overlap dialing
(Reported by N A)
* ASTERISK-30319 - Add BYE Reason support for SIP
(Reported by Igor Goncharovsky)
* ASTERISK-30180 - app_broadcast: Add a channel audio
multicasting application
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
(Reported by AvayaXAsterisk)
* ASTERISK-30354 - chan_iax2: Lack of formats prior to
receiving voice frames causes jitterbuffer to stall
(Reported by N A)
* ASTERISK-30162 - when chan_iax is used to relay calls, no
ringing indication is played
(Reported by Jaco Kroon)
* ASTERISK-30424 - pjproject_bundled: cross-compilation broken
when ssl autodetected
(Reported by Nick French)
* ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
multi-homed
(Reported by cmaj)
* ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13
(Reported by Ross Beer)
* ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
(Reported by Sean Bright)
* ASTERISK-30406 - pbx_ael: Global variables are not expanded.
(Reported by Sean Bright)
* ASTERISK-29604 - ari: Segfault with lots of calls
(Reported by Danila Evgrafov)
* ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
g722 after MES changes
(Reported by George Joseph)
* ASTERISK-30345 - loader.c: Modules that decline to load
cannot be reloaded
(Reported by N A)
* ASTERISK-30379 - http: fix NULL pointer dereference while
enable_status on TLS-only
(Reported by Boris P. Korzun)
* ASTERISK-30375 - res_http_media_cache: Crash when URL has no
path component.
(Reported by Sean Bright)
* ASTERISK-30351 - manager: Originate variables are not added
when setvar used in manager.conf
(Reported by Sebastian
Gutierrez)
* ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
when they shouldn't be
(Reported by Joshua C. Colp)
* ASTERISK-30367 - pbx: Fix outdated channel snapshots with pbx_exec
(Reported by N A)
* ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
for extension, callerid supplement executed too late
(Reported by Oleg)
* ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
used when moh_passthrough has call on hold
(Reported by Benjamin Keith Ford)
* ASTERISK-30240 - app voicemail odbc build error with gcc 11.1
(Reported by Michael Bradeen)
* ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to endpoint
(Reported by Yury Kirsanov)
* ASTERISK-30198 - Error `Too many open files` occurs after
about ~8000 calls when using mixmonitor
(Reported by Julien Alie)
Improvements made in this release:
-----------------------------------
* ASTERISK-30411 - app_read: add option to include terminating
digit on empty, terminated strings
(Reported by Michael Bradeen)
* ASTERISK-30405 - app_directory: Add 's' option to skip
channel call
(Reported by Michael Bradeen)
* ASTERISK-30422 - app_senddtmf: add the option for senddtmf to answer
(Reported by Michael Bradeen)
* ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
(Reported by Stanislav Abramenkov)
* ASTERISK-30404 - app_directory: Add reading directory
configuration from custom file
(Reported by Michael Bradeen)
* ASTERISK-29913 - func_json: Adds multi-level and array
parsing to JSON_DECODE
(Reported by N A)
* ASTERISK-30353 - func_frame_trace: Print text for text frames
(Reported by N A)
* ASTERISK-30361 - json.h: Add missing
ast_json_object_real_get
(Reported by N A)
* ASTERISK-30280 - Create capability to assign a Media
Experience Score to RTP streams
(Reported by George Joseph)
* ASTERISK-30332 - func_callerid: Warn if invalid redirecting
reason provided
(Reported by N A)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
Thank you for your continued support of Asterisk!
[asterisk-announce] Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available
The Asterisk Development Team would like to announce the release of
Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues reported by the
community and would have not been possible without your participation.Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------
[ASTERISK-30103 <https://issues.asterisk.org/jira/browse/ASTERISK-30103>]
chan_ooh323 vulnerability in calling/called party IE (Reported By: Michael
Bradeen)
[ASTERISK-30176 <https://issues.asterisk.org/jira/browse/ASTERISK-30176>]
GetConfig can read files outside of Asterisk (Reported By: shawty)
[ASTERISK-30244 <https://issues.asterisk.org/jira/browse/ASTERISK-30244>]
Occasional crash when TCP/TLS connection terminated and subscription
persistence is removed (Reported By: nappsoft)
[ASTERISK-30338 <https://issues.asterisk.org/jira/browse/ASTERISK-30338>]
Backport 2.13 security fixes from pjproject
[asterisk-announce] Asterisk 18.15.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.15.0.
The release of Asterisk 18.15.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-30037 - Add test support to calling external processes
(Reported by Philip Prindeville)
* ASTERISK-30161 - locks: add AMI event for deadlock
(Reported by N A)
* ASTERISK-30211 - app_confbridge: Add end_marked_any option
(Reported by N A)
* ASTERISK-30186 - res_pjsip: Add support for reloading TLS
certificate and key information
(Reported by Joshua C. Colp)
* ASTERISK-29899 - features: Add advanced transfer initiation options
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30235 - res_crypto and tests: Memory issues and and
uninitialized variable error
(Reported by George Joseph)
* ASTERISK-30234 - res_geolocation: ...may be used
uninitialized error in geoloc_config.c
(Reported by George Joseph)
* ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
a Segmentation Fault
(Reported by Dan Cropp)
* ASTERISK-30135 - [res_musiconhold] Allows the moh only for
the answered call
(Reported by sungtae kim)
* ASTERISK-26894 - pjsip should support tel uri scheme
(Reported by Gergely D½½ms½½di)
* ASTERISK-30210 - func_frame_trace: Channel masquerade
triggers assertion
(Reported by N A)
* ASTERISK-30190 - res_geolocation: GEOLOC_PROFILE isn't
returning correct values on incoming channel
(Reported by George Joseph)
* ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is broken.
(Reported by Alexander Traud)
* ASTERISK-30192 - res_tonedetect: fix typo for frametype
(Reported by N A)
* ASTERISK-29453 - alembic: incoming_call_offer_pref and
outgoing_call_offer_pref missing in "ps_endpoints" table
(Reported by Daniel Th½½men)
* ASTERISK-26826 - testsuite: Add support for Python 3
(Reported by Joshua C. Colp)
* ASTERISK-30167 - res_geolocation: Refactor for issues found
by users
(Reported by George Joseph)
* ASTERISK-28422 - Memory Leak in Confbridge menu
(Reported by Ted G)
* ASTERISK-29917 - ami: FilterList action doesn't exist
(Reported by N A)
* ASTERISK-30018 - app_meetme: MeetmeList AMI event not documented
(Reported by Michael Cargile)
* ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
(Reported by Michael Cargile)
* ASTERISK-30151 - Documentation doesn't include info about
"field", a 3rd required parameter.
(Reported by Chris Young)
Improvements made in this release:
-----------------------------------
* ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
scope trace debugs to DEBUG level
(Reported by N A)
* ASTERISK-30178 - extend user_eq_phone behavior to local uri's
(Reported by Michael Bradeen)
* ASTERISK-30046 - Reimplement res/res_crypto.c internals with
EVP_PKEY interface to Openssl API's
(Reported by Philip Prindeville)
* ASTERISK-30045 - Add test coverage to res/res_crypto.c functionality
(Reported by Philip Prindeville)
* ASTERISK-30185 - res_geolocation: Allow location parameters
to be specified in profiles
(Reported by George Joseph)
* ASTERISK-30177 - res_geolocation: Add option to suppress
empty elements
(Reported by George Joseph)
* ASTERISK-30182 - res_geolocation: Add built-in profiles to
use in fully dynamic configurations
(Reported by George Joseph)
* ASTERISK-29906 - [patch] update RLS to reflect the changes to
the lists
(Reported by Alexei Gradinari)
* ASTERISK-30163 - general: fix minor formatting issues
(Reported by N A)
* ASTERISK-30164 - chan_iax2: Add missing option documentation
(Reported by N A)
* ASTERISK-30153 - logger: Improve log levels
(Reported by N A)
* ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql reference
(Reported by N A)
* ASTERISK-30159 - general: Remove obsolete SVN references
(Reported by N A)
[asterisk-announce] Asterisk 18.14.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.14.0.
The release of Asterisk 18.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-30128 - Create PJSIP interface module for Geolocation
(Reported by George Joseph)
* ASTERISK-30127 - Create core Geolocation capability for Asterisk
(Reported by George Joseph)
* ASTERISK-30089 - general: fix typos
(Reported by N A)
* ASTERISK-30050 - Upgrade Asterisk to bundled pjproject 2.12.1
(Reported by Stanislav Abramenkov)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30167 - res_geolocation: Refactor for issues found
by users
(Reported by George Joseph)
* ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
(Reported by N A)
* ASTERISK-29905 - OSX: bininstall launchd issue on
cross-platfrom build
(Reported by Sergey V. Lobanov)
* ASTERISK-30137 - manager: Global disabled event filtered is incomplete
(Reported by N A)
* ASTERISK-30109 - res_pjsip: no contact-status AMI event on
register of prune-on-boot contact that uses the same URI as
before Asterisk restart
(Reported by Michael Neuhauser)
* ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
honor presentation
(Reported by N A)
* ASTERISK-30126 - Spelling mistake in
configs/samples/queues.conf.sample
(Reported by Sam Banks)
* ASTERISK-30029 - build: Git security vulnerability fix is sad
with our accessing git as root during "make install"
(Reported by Joshua C. Colp)
* ASTERISK-29907 - res_pjsip, app_confbridge: Video call
through ConfBridge with normal endpoints causes infinite
loop/crash
(Reported by N A)
* ASTERISK-30138 - Compile failure in
res_geolocation/geoloc_eprofile.c when optimization is enabled
(Reported by George Joseph)
* ASTERISK-30096 - cel_odbc: Column type 9 (field
'cdr:cel:eventtime') is unsupported at this time
(Reported by Morvai Szabolcs)
* ASTERISK-30083 - chan_iax2: Optional dependency on
openssl/res_crypto is now mandatory
(Reported by Dmitry Melekhov)
* ASTERISK-30099 - test_aeap_transport: transport_connect_fail
sporadically causes failure
(Reported by Kevin Harwell)
* ASTERISK-30123 - features: Update automixmon documentation to
reflect reality
(Reported by Trevor Peirce)
* ASTERISK-30117 - pbx_lua: Remove compiler warnings
(Reported by Boris P. Korzun)
* ASTERISK-30101 - res_prometheus: Optional load
res_pjsip_outbound_registration.so
(Reported by Boris P. Korzun)
* ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
inconsistent for busy
(Reported by N A)
* ASTERISK-30001 - db: Removing nonexistent entries shows
"Database entry removed"
(Reported by N A)
* ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
outbound dials
(Reported by N A)
* ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
calendars no longer work
(Reported by N A)
* ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
with remote console
(Reported by N A)
* ASTERISK-30072 - res_pjsip: allow TLS verification of
wildcard cert-bearing servers
(Reported by Kevin Harwell)
* ASTERISK-30075 - say: Abort if channel hangs up during playback
(Reported by N A)
New Features made in this release:
-----------------------------------
* ASTERISK-30136 - db: Add AMI action to retrieve all keys
beginning with a prefix
(Reported by N A)
* ASTERISK-30000 - chan_dahdi: Add POLARITY function
(Reported by N A)
* ASTERISK-30062 - cli: Add CLI command to execute a dialplan app
(Reported by N A)
* ASTERISK-29999 - pjsip: Get information from 200 OK INVITE
reply headers
(Reported by Jos½½ Lopes)
* ASTERISK-30061 - pbx: Add pbx helper application
(Reported by N A)
[asterisk-announce] Asterisk 18.13.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.13.0.
The release of Asterisk 18.13.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-29906 - [patch] update RLS to reflect the changes to the lists
(Reported by Alexei Gradinari)
* ASTERISK-29891 - [patch] provide a display name for RLS subscriptions
(Reported by Alexei Gradinari)
* ASTERISK-30090 - xmldocs: Use example tags for examples
(Reported by N A)
* ASTERISK-30086 - res_parking: Warn when invalid parking space requested
(Reported by N A)
* ASTERISK-30058 - Evaluate dialplan functions and variables in agi exec
(Reported by Shloime Rosenblum)
* ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
Call-ID for chan_sip/chan_pjsip) in ARI channel resource
(Reported by Moritz Fain)
* ASTERISK-29845 - res_pjsip_outbound_registration: Show time
remaining until registration lapses
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30097 - console: Recent documentation changes for
connecting to remote console are inconsistent
(Reported by Matthias Hensler)
* ASTERISK-30043 - Wrong party is disconnected when
hook-flashing on 3-way bridge
(Reported by Josh Alberts)
* ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
"timers=always" is specified in pjsip.conf
(Reported by Ray Crumrine)
* ASTERISK-30092 - DateTime application: wrong inflection for
one o'clock in German
(Reported by Christof Efkemann)
* ASTERISK-30064 - pbx: iax2 switch causes crash due to
deadlock and assertion
(Reported by N A)
* ASTERISK-29981 - res_calendar: Asterisk crashes when
starting, and will not run
(Reported by N A)
* ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
creates unstable system
(Reported by N A)
* ASTERISK-30051 - res_pjsip: No video after un-hold with
moh_passthrough=yes
(Reported by Maximilian Fridrich)
* ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
in PJSIP NOTIFY event: dialog XML body
(Reported by Marco Paland)
* ASTERISK-30059 - menuselect: libxml include fails under Gentoo
(Reported by waltermoeller)
* ASTERISK-30060 - loader: format warnings in dev mode
(Reported by N A)
* ASTERISK-30065 - pjsip: Open Websocket connection is not
reused for outgoing requests
(Reported by LA)
* ASTERISK-30042 - res_pjsip_transport_websocket: Registration
over websocket returns a rewritten contact
(Reported by Thomas Guebels)
* ASTERISK-29993 - chan_dahdi: Operator control option borks
both lines involved on callee disconnect
(Reported by N A)
* ASTERISK-30044 - GCC 12 issues
(Reported by George Joseph)
New Features made in this release:
-----------------------------------
* ASTERISK-30063 - app_voicemail: Add option to prevent
deletion of messages
(Reported by N A)
* ASTERISK-29965 - res_pjsip_outbound_registration: Make max
registration delay configurable
(Reported by N A)
* ASTERISK-30087 - res_parking: Add music on hold override option
(Reported by N A)
* ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS function
(Reported by N A)
Thank you for your continued support of Asterisk!
[asterisk-announce] Asterisk 18.12.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.12.1.
The release of Asterisk 18.12.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30065 - pjsip: Open Websocket connection is not
reused for outgoing requests
(Reported by LA)
[asterisk-announce] Asterisk 18.12.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.12.0.
The release of Asterisk 18.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities
(Reported by Clint Ruoho)
* ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a terminating \
(Reported by Leandro Dardini)
* ASTERISK-29872 - res_stir_shaken: Resource exhaustion with large files
(Reported by Benjamin Keith Ford)
New Features made in this release:
-----------------------------------
* ASTERISK-29931 - Option to allow a user to not hear the join
sound on enter but everyone else can
(Reported by Michael Cargile)
* ASTERISK-29968 - func_db: Add a function to return
cardinality of keys at prefix
(Reported by N A)
* ASTERISK-29486 - Hint-like extension value lookup function
without device state
(Reported by N A)
* ASTERISK-29941 - chan_pjsip: Add ability to send flash events
(Reported by N A)
* ASTERISK-29820 - cli: Add command to evaluate a function
(Reported by N A)
* ASTERISK-29876 - app_queue: Add music on hold option
(Reported by N A)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29655 - res_pjsip_session: No video to caller if no
camera available
(Reported by Michael Auracher)
* ASTERISK-29638 - res_pjsip_session: No video after early media
(Reported by Michael Auracher)
* ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
when Picking Up Dahdi Call On Hold
(Reported by Josh Alberts)
* ASTERISK-29990 - chan_dahdi: adding ring cadences is not
idempotent on dahdi restart
(Reported by N A)
* ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
encryption with missing secrets
(Reported by N A)
* ASTERISK-29728 - menuselect: Disabled by default modules that
are enabled are always recompiled
(Reported by N A)
* ASTERISK-30002 - app_meetme: Don't erroneously set global
variables when channel is NULL
(Reported by N A)
* ASTERISK-29994 - chan_dahdi: Round robin array size is too
small for max number of groups
(Reported by N A)
* ASTERISK-22246 - Asterisk's "T" flag is ignored when used
with "r" or "R" flags. (documentation bug)
(Reported by Rusty Newton)
* ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
for "disable console colorization"
(Reported by Sebastian Gutierrez)
* ASTERISK-29843 - Session timers get removed on UPDATE
(Reported by Mark Petersen)
* ASTERISK-29943 - file.c: seeking to negative file offset is
not prevented
(Reported by N A)
* ASTERISK-29955 - chan_sip: SIP route header is missing on UPDATE
(Reported by Mark Petersen)
* ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
even if early_media already enabled
(Reported by Mark Petersen)
* ASTERISK-29948 - iostream: Infinite TCP timeout writing data
(Reported by N A)
* ASTERISK-29253 - Incorrect bridging on transfer
(Reported by Yury Kirsanov)
* ASTERISK-30006 - res_pjsip: UDP transport does not work when
async_operations is greater than 1
(Reported by Ross Beer)
* ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
functionality not enabled
(Reported by Claude Diderich)
* ASTERISK-30021 - ast_variable_list_replace_variable uses
variable with new keyword
(Reported by Jasper Hafkenscheid)
* ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
database columns
(Reported by Gregory Massel)
* ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
of SDP attributes
(Reported by Josh Hogan)
* ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
Disconnecting channel for lack of RTP activity
(Reported by Dmitriy Serov)
* ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
lack of RTP activity in one way sessions
(Reported by Boris P. Korzun)
* ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
(Reported by LA)
* ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
(Reported by Daniel Bonazzi)
* ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
context (AST_PBX_MAX_STACK - 1)
(Reported by Tzafrir Cohen)
* ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
wget isn't available
(Reported by Stefan Ruijsenaars)
* ASTERISK-29988 - REGRESSION: The build process is requiring
xmllint or xmlstarlet ro be installed when it shouldn't
(Reported by George Joseph)
* ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
show netstats printout
(Reported by N A)
* ASTERISK-29939 - agi: Fix xmldoc bug with set music
(Reported by N A)
* ASTERISK-28891 - documentation: AGICommand_set+music
documentation arguments displayed incorreclty
(Reported by Jonathan Harris)
* ASTERISK-29048 - chan_iax2: "iax2 show registry" shows host
for perceived
(Reported by David Herselman)
* ASTERISK-29674 - Adjust for 64bit time_t
(Reported by Andre Heider)
* ASTERISK-29961 - RLS: domain part of 'uri' list attribute
mismatch with SUBSCRIBE request
(Reported by Alexei Gradinari)
* ASTERISK-29928 - logging messages truncated when using MUSL runtime
(Reported by Philip Prindeville)
* ASTERISK-29960 - ari: Retrieving stored recording can returns
wrong file
(Reported by Arix)
* ASTERISK-29950 - SayNumber can handle '01' to '07', but not
'08' or '09'
(Reported by Jim Van Meggelen)
Improvements made in this release:
-----------------------------------
* ASTERISK-24827 - Missing documentation for chan_dahdi dial
string ring cadences
(Reported by Scott Griepentrog)
* ASTERISK-29940 - general: Add since tags to xmldocs
(Reported by N A)
* ASTERISK-29726 - Add Asterisk External Application Protocol
(AEAP) implementation
(Reported by Kevin Harwell)
* ASTERISK-29951 - app_mf, app_sf: Return -1 on hangup
(Reported by N A)
* ASTERISK-29954 - app_meetme: Emit warning if conference not found
(Reported by N A)
* ASTERISK-29351 - Qualify pjproject 2.12 for Asterisk
(Reported by George Joseph)
* ASTERISK-29976 - Should Readme include information about
install_prereq script?
(Reported by Marcel Wagner)
* ASTERISK-29970 - Use pkg-config to find libxml2 headers and libraries
(Reported by Hugh McMaster)
* ASTERISK-29980 - build: External binary modules don't use https
(Reported by INVADE International Ltd.)
* ASTERISK-25716 - Documentation: Document explanations and
examples for possible values of DIALSTATUS
(Reported by Rusty Newton)
* ASTERISK-29967 - pbx_builtins: Add missing documentation
(Reported by N A)
[asterisk-announce] Asterisk 18.11.3 Now Available
Asterisk Development Team asteriskteam at digium.com
Tue Apr 26 12:09:50 CDT 2022
The Asterisk Development Team would like to announce the release of Asterisk 18.11.3.
The release of Asterisk 18.11.3 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following issue is resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
functionality not enabled
(Reported by Claude Diderich)
[asterisk-announce] Asterisk 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14 Now Available (Security)
The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
The following security vulnerabilities were resolved in these versions:
* AST-2022-001: res_stir_shaken: resource exhaustion with large files
When using STIR/SHAKEN, it½½½s possible to download files that are not
certificates. These files could be much larger than what you would expect to
download.
* AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header
When using STIR/SHAKEN, it½½½s possible to send arbitrary requests like GET to
interfaces such as localhost using the Identity header.
* AST-2022-003: func_odbc: Possible SQL Injection
Some databases can use backslashes to escape certain characters, such as
backticks. If input is provided to func_odbc which includes backslashes it is
possible for func_odbc to construct a broken SQL query and the SQL query to
fail.
[asterisk-announce] Asterisk 18.11.1 Now Available
Asterisk Development Team asteriskteam at digium.com
Tue Mar 29 19:15:43 CDT 2022
The Asterisk Development Team would like to announce the release of Asterisk 18.11.1.
The release of Asterisk 18.11.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
wget isn't available
(Reported by Stefan Ruijsenaars)
* ASTERISK-29988 - REGRESSION: The build process is requiring
xmllint or xmlstarlet ro be installed when it shouldn't
(Reported by George Joseph)
[asterisk-announce] Asterisk 18.11.0 Now Available
Asterisk Development Team asteriskteam at digium.com
Thu Mar 24 09:06:03 CDT 2022
The Asterisk Development Tea…- Loading branch information