The Asterisk Development Team would like to announce
release candidate 1 of Certified asterisk-20.7-cert1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert1-rc1
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

Repository: https://github.com/asterisk/asterisk
Tag: certified-20.7-cert1-rc1

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-certified-20.7-cert1-rc1

Summary:

Initial commit for certified-20.7
res_pjsip_stir_shaken.c: Add checks for missing parameters
app_dial: Add dial time for progress/ringing.
app_voicemail: Properly reinitialize config after unit tests.
app_queue.c : fix "queue add member" usage string
app_voicemail: Allow preventing mark messages as urgent.
res_pjsip: Use consistent type for boolean columns.
attestation_config.c: Use ast_free instead of ast_std_free
Makefile: Add stir_shaken/cache to directories created on install
Stir/Shaken Refactor
alembic: Synchronize alembic heads between supported branches.
translate.c: implement new direct comp table mode
README.md: Removed outdated link
strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
res_rtp_asterisk.c: Correct coefficient in MOS calculation.
dsp.c: Fix and improve potentially inaccurate log message.
pjsip show channelstats: Prevent possible segfault when faxing
Reduce startup/shutdown verbose logging
configure: Rerun bootstrap on modern platform.
Upgrade bundled pjproject to 2.14.
app_speech_utils.c: Allow partial speech results.
utils: Make behavior of ast_strsep* match strsep.
app_chanspy: Add 'D' option for dual-channel audio
app_if: Fix next priority calculation.
res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
BuildSystem: Bump autotools versions on OpenBSD.
main/utils: Simplify the FreeBSD ast_get_tid() handling
res_pjsip_session.c: Correctly format SDP connection addresses.
rtp_engine.c: Correct sample rate typo for L16/44100.
manager.c: Fix erroneous reloads in UpdateConfig.
res_calendar_icalendar: Print iCalendar error on parsing failure.
app_confbridge: Don't emit warnings on valid configurations.
app_voicemail: add NoOp alembic script to maintain sync
chan_dahdi: Allow MWI to be manually toggled on channels.
chan_rtp.c: MulticastRTP missing refcount without codec option
chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
func_frame_trace: Add CLI command to dump frame queue.
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 cha..
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.
Update config.yml
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 m..
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
res_rtp_asterisk: Fix regression issues with DTLS client check
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.
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
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."
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.
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 par..
app_voicemail: add CLI commands for message manipulation
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using rtp->themssrc_valid i..
sig_analog: Allow immediate fake ring to be suppressed.
app.h: Move declaration of ast_getdata_result before its first use
doc: Remove obsolete CHANGES-staging and UPGRADE-staging
app_voicemail: fix imap compilation errors
res_musiconhold: avoid moh state access on unlocked chan
utils: add lock timestamps for DEBUG_THREADS
rest-api: Updates for new documentation site
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
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
indications: logging changes
callerid: Allow specifying timezone for date/time.
logrotate: Fix duplicate log entries.
chan_pjsip: Allow topology/session refreshes in early media state
chan_dahdi: Fix broken hidecallerid setting.
asterisk.c: Fix option warning for remote console.
configure: fix test code to match gethostbyname_r prototype.
res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
res_sorcery_memory_cache.c: Fix memory leak
xml.c: Process XML Inclusions recursively.
apply_patches: Use globbing instead of file/sort.
apply_patches: Sort patch list before applying
pjsip: Upgrade bundled version to pjproject 2.13.1
Set up new ChangeLogs directory
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. (#57)
res_pjsip: mediasec: Add Security-Client headers after 401
LICENSE: Update link to trademark policy.
chan_dahdi: Add dialmode option for FXS lines.
Initial GitHub PRs
Initial GitHub Issue Templates
pbx_dundi: Fix PJSIP endpoint configuration check.
Revert "app_queue: periodic announcement configurable start time."
res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
pbx_dundi: Add PJSIP support.
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
res_calendar: output busy state as part of show calendar.
loader.c: Minor module key check simplification.
ael: Regenerate lexers and parsers.
bridge_builtin_features: add beep via touch variable
res_mixmonitor: MixMonitorMute by MixMonitor ID
format_sln: add .slin as supported file extension
res_agi: RECORD FILE plays 2 beeps.
func_json: Fix JSON parsing issues.
app_senddtmf: Add SendFlash AMI action.
app_dial: Fix DTMF not relayed to caller on unanswered calls.
configure: fix detection of re-entrant resolver functions
cli: increase channel column width
app_queue: periodic announcement configurable start time.
make_version: Strip svn stuff and suppress ref HEAD errors
res_http_media_cache: Introduce options and customize
main/iostream.c: fix build with libressl
contrib: rc.archlinux.asterisk uses invalid redirect.
res_pjsip_pubsub: subscription cleanup changes
Revert "pbx_ael: Global variables are not expanded."
res_pjsip: Replace invalid UTF-8 sequences in callerid name
test.c: Avoid passing -1 to FD_* family of functions.
chan_iax2: Fix jitterbuffer regression prior to receiving audio.
test_crypto.c: Fix getcwd(…) build error.
pjproject_bundled: Fix cross-compilation with SSL libs.
app_read: Add an option to return terminator on empty digits.
res_phoneprov.c: Multihomed SERVER cache prevention
app_directory: Add a 'skip call' option.
app_senddtmf: Add option to answer target channel.
res_pjsip: Prevent SEGV in pjsip_evsub_send_request
app_queue: Minor docs and logging fixes for UnpauseQueueMember.
app_queue: Reset all queue defaults before reload.
res_pjsip: Upgraded bundled pjsip to 2.13
doxygen: Fix doxygen errors.
app_signal: Add signaling applications
app_directory: add ability to specify configuration file
func_json: Enhance parsing capabilities of JSON_DECODE
res_stasis_snoop: Fix snoop crash
pbx_ael: Global variables are not expanded.
res_pjsip_session: Add overlap_context option.
app_playback.c: Fix PLAYBACKSTATUS regression.
res_rtp_asterisk: Don't use double math to generate timestamps
format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
res_pjsip_rfc3326: Add SIP causes support for RFC3326
res_rtp_asterisk: Asterisk Media Experience Score (MES)
Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"
loader: Allow declined modules to be unloaded.
app_broadcast: Add Broadcast application
func_frame_trace: Print text for text frames.
json.h: Add ast_json_object_real_get.
manager: Fix appending variables.
res_pjsip_transport_websocket: Add remote port to transport
http.c: Fix NULL pointer dereference bug
res_http_media_cache: Do not crash when there is no extension
res_rtp_asterisk: Asterisk Media Experience Score (MES)
pbx_app: Update outdated pbx_exec channel snapshots.
res_pjsip_session: Use Caller ID for extension matching.
res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
app_voicemail_odbc: Fix string overflow warning.
func_callerid: Warn about invalid redirecting reason.
res_pjsip: Fix path usage in case dialing with '@'
streams: Ensure that stream is closed in ast_stream_and_wait on error
app_sendtext: Remove references to removed applications.
res_geoloc: fix NULL pointer dereference bug
res_pjsip_aoc: Don't assume a body exists on responses.
app_if: Fix format truncation errors.
manager: AOC-S support for AOCMessage
res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
ari: Destroy body variables in channel create.
app_voicemail: Fix missing email in msg_create_from_file.
res_pjsip: Fix typo in from_domain documentation
res_hep: Add support for named capture agents.
app_if: Adds conditional branch applications
res_pjsip_session.c: Map empty extensions in INVITEs to s.
res_pjsip: Update contact_user to point out default
res_adsi: Fix major regression caused by media format rearchitecture.
res_pjsip_header_funcs: Add custom parameter support.
func_presencestate: Fix invalid memory access.
sig_analog: Fix no timeout duration.
xmldoc: Allow XML docs to be reloaded.
rtp_engine.h: Update examples using ast_format_set.
app_mixmonitor: Add option to use real Caller ID for voicemail.
pjproject: 2.13 security fixes
pjsip_transport_events: Fix possible use after free on transport
manager: prevent file access outside of config dir
ooh323c: not checking for IE minimum length
pbx_builtins: Allow Answer to return immediately.
chan_dahdi: Allow FXO channels to start immediately.
core & res_pjsip: Improve topology change handling.
sla: Prevent deadlock and crash due to autoservicing.
Build system: Avoid executable stack.
func_json: Fix memory leak.
test_json: Remove duplicated static function.
res_agi: Respect "transmit_silence" option for "RECORD FILE".
app_mixmonitor: Add option to delete files on exit.
manager: Update ModuleCheck documentation.
file.c: Don't emit warnings on winks.
runUnittests.sh: Save coredumps to proper directory
app_stack: Print proper exit location for PBXless channels.
chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer
res_pjsip: prevent crash on websocket disconnect
tcptls: Prevent crash when freeing OpenSSL errors.
res_pjsip_outbound_registration: Allow to use multiple proxies for registration
tests: Fix compilation errors on 32-bit.
res_pjsip: return all codecs on a re-INVITE without SDP
res_pjsip_notify: Add option support for AMI.
res_pjsip_logger: Add method-based logging option.
Dialing API: Cancel a running async thread, may not cancel all calls
chan_dahdi: Fix unavailable channels returning busy.
res_pjsip_pubsub: Prevent removing subscriptions.
say: Don't prepend ampersand erroneously.
res_crypto: handle unsafe private key files
audiohook: add directional awareness
cdr: Allow bridging and dial state changes to be ignored.
res_tonedetect: Add ringback support to TONE_DETECT.
chan_dahdi: Resolve format truncation warning.
res_crypto: don't modify fname in try_load_key()
res_crypto: use ast_file_read_dirs() to iterate
res_geolocation: Update wiki documentation
res_pjsip: Add mediasec capabilities.
res_prometheus: Do not crash on invisible bridges
res_pjsip_geolocation: Change some notices to debugs.
db: Fix incorrect DB tree count for AMI.
func_logic: Don't emit warning if both IF branches are empty.
features: Add no answer option to Bridge.
app_bridgewait: Add option to not answer channel.
app_amd: Add option to play audio during AMD.
test: initialize capture structure before freeing
func_export: Add EXPORT function
res_pjsip: Add 100rel option "peer_supported".
func_scramble: Fix null pointer dereference.
manager: be more aggressive about purging http sessions.
func_strings: Add trim functions.
res_crypto: Memory issues and uninitialized variable errors
res_geolocation: Fix issues exposed by compiling with -O2
res_crypto: don't complain about directories
res_pjsip: Add user=phone on From and PAID for usereqphone=yes
res_geolocation: Fix segfault when there's an empty element
res_musiconhold: Add option to not play music on hold on unanswered channels
res_pjsip: Add TEL URI support for basic calls.
res_crypto: Use EVP API's instead of legacy API's
test: Add coverage for res_crypto
res_crypto: make keys reloadable on demand for testing
test: Add test coverage for capture child process output
main/utils: allow checking for command in $PATH
test: Add ability to capture child process output
res_crypto: Don't load non-regular files in keys directory
func_frame_trace: Remove bogus assertion.
lock.c: Add AMI event for deadlocks.
app_confbridge: Add end_marked_any option.
pbx_variables: Use const char if possible.
res_geolocation: Add two new options to GEOLOC_PROFILE
res_geolocation: Allow location parameters on the profile object
res_geolocation: Add profile parameter suppress_empty_ca_elements
res_geolocation: Add built-in profiles
res_pjsip_sdp_rtp: Skip formats without SDP details.
cli: Prevent assertions on startup from bad ao2 refs.
pjsip: Add TLS transport reload support for certificate and key.
res_tonedetect: Fix typos referring to wrong variables.
alembic: add missing ps_endpoints columns
chan_dahdi.c: Resolve a format-truncation build warning.
res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update
channel.h: Remove redundant declaration.
features: Add transfer initiation options.
CI: Fixing path issue on venv check
CI: use Python3 virtual environment
general: Very minor coding guideline fixes.
res_geolocation: Address user issues, remove complexity, plug leaks
chan_iax2: Add missing options documentation.
app_confbridge: Fix memory leak on updated menu options.
Geolocation: Wiki Documentation
manager: Remove documentation for nonexistent action.
general: Improve logging levels of some log messages.
cdr.conf: Remove obsolete app_mysql reference.
general: Remove obsolete SVN references.
app_confbridge: Add missing AMI documentation.
app_meetme: Add missing AMI documentation.
func_srv: Document field parameter.
pbx_functions.c: Manually update ast_str strlen.
build: fix bininstall launchd issue on cross-platform build
db: Add AMI action to retrieve DB keys at prefix.
manager: Fix incomplete filtering of AMI events.
Update defaultbranch to 20
res_pjsip: delay contact pruning on Asterisk start
chan_dahdi: Fix buggy and missing Caller ID parameters
queues.conf.sample: Correction of typo
chan_dahdi: Add POLARITY function.
Makefile: Avoid git-make user conflict
app_confbridge: Always set minimum video update interval.
pbx.c: Simplify ast_context memory management.
geoloc_eprofile.c: Fix setting of loc_src in set_loc_src()
Geolocation: chan_pjsip Capability Preview
Geolocation: Core Capability Preview
general: Fix various typos.
cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type
chan_iax2: Allow compiling without OpenSSL.
websocket / aeap: Handle poll() interruptions better.
res_cliexec: Add dialplan exec CLI command.
features: Update documentation for automon and automixmon
Geolocation: Base Asterisk Prereqs
pbx_lua: Remove compiler warnings
res_pjsip_header_funcs: Add functions PJSIP_RESPONSE_HEADER and PJSIP_RESPONSE..
res_prometheus: Optional load res_pjsip_outbound_registration.so
app_dial: Fix dial status regression.
db: Notify user if deleted DB entry didn't exist.
cli: Fix CLI blocking forever on terminating backslash
app_dial: Propagate outbound hook flashes.
res_calendar_icalendar: Send user agent in request.
say: Abort play loop if caller hangs up.
res_pjsip: allow TLS verification of wildcard cert-bearing servers
pbx: Add helper function to execute applications.
pjsip: Upgrade bundled version to pjproject 2.12.1
asterisk.c: Fix incompatibility warnings for remote console.
test_aeap_transport: disable part of failing unit test
sig_analog: Fix broken three-way conferencing.
app_voicemail: Add option to prevent message deletion.
res_parking: Add music on hold override option.
xmldocs: Improve examples.
res_pjsip_outbound_registration: Make max random delay configurable.
res_pjsip: Actually enable session timers when timers=always
res_pjsip_pubsub: delete scheduled notification on RLS update
res_pjsip_pubsub: XML sanitized RLS display name
app_sayunixtime: Use correct inflection for German time.
chan_iax2: Prevent deadlock due to duplicate autoservice.
loader: Prevent deadlock using tab completion.
res_calendar: Prevent assertion if event ends in past.
res_parking: Warn if out of bounds parking spot requested.
chan_pjsip: Only set default audio stream on hold.
res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI
res_agi: Evaluate dialplan functions and variables in agi exec if enabled
ast_pkgconfig.m4: AST_PKG_CONFIG_CHECK() relies on sed.
ari: expose channel driver's unique id to ARI channel resource
loader.c: Use portable printf conversion specifier for int64.
res_pjsip_transport_websocket: Also set the remote name.
res_pjsip_transport_websocket: save the original contact host
res_pjsip_outbound_registration: Show time until expiration
app_confbridge: Add function to retrieve channels.
chan_dahdi: Fix broken operator mode clearing.
GCC12: Fixes for 16+
GCC12: Fixes for 18+. state_id_by_topic comparing wrong value
core_unreal: Flip stream direction of second channel.
chan_dahdi: Document dial resource options.
chan_dahdi: Don't allow MWI FSK if channel not idle.
apps/confbridge: Added hear_own_join_sound option to control who hears sound_j..
chan_dahdi: Don't append cadences on dahdi restart.
chan_iax2: Prevent crash if dialing RSA-only call without outkey.
menuselect: Don't erroneously recompile modules.
app_meetme: Don't erroneously set global variables.
asterisk.c: Warn of incompatibilities with remote console.
func_db: Add function to return cardinality at prefix
chan_dahdi: Fix insufficient array size for round robin.
chan_sip.c Session timers get removed on UPDATE
func_evalexten: Extension evaluation function.
file.c: Prevent formats from seeking negative offsets.
chan_pjsip: Add ability to send flash events.
cli: Add command to evaluate dialplan functions.
documentation: Adds versioning information.
samples: Remove obsolete sample configs.
chan_pjsip: add allow_sending_180_after_183 option
chan_sip: SIP route header is missing on UPDATE
manager: Terminate session on write error.
bridge_simple.c: Unhold channels on join simple bridge.
res_aeap & res_speech_aeap: Add Asterisk External Application Protocol
app_dial: Flip stream direction of outgoing channel.
res_pjsip_stir_shaken.c: Fix enabled when not configured.
res_pjsip: Always set async_operations to 1.
config.h: Don't use C++ keywords as argument names.
cdr_adaptive_odbc: Add support for SQL_DATETIME field type.
pjsip: Increase maximum number of format attributes.
AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header.
AST-2022-001 - res_stir_shaken/curl: Limit file size and check start.
func_odbc: Add SQL_ESC_BACKSLASHES dialplan function.
app_mf, app_sf: Return -1 if channel hangs up.
app_queue: Add music on hold option to Queue.
app_meetme: Emit warning if conference not found.
build: Remove obsolete leftover build references.
res_pjsip_header_funcs: wrong pool used tdata headers
deprecation cleanup: remove leftover files
pjproject: Update bundled to 2.12 release.
pbx.c: Warn if there are too many includes in a context.
Makefile: Disable XML doc validation
make_xml_documentation: Remove usage of get_sourceable_makeopts
chan_iax2: Fix spacing in netstats command
openssl: Supress deprecation warnings from OpenSSL 3.0
documentation: Add information on running install_prereq script in readme
chan_iax2: Fix perceived showing host address.
res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity
configure.ac: Use pkg-config to detect libxml2
time: add support for time64 libcs
res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request
app_dial: Document DIALSTATUS return values.
stasis_recording: Perform a complete match on requested filename.
download_externals: Use HTTPS for downloads
conversions.c: Specify that we only want to parse decimal numbers.
logger: workaround woefully small BUFSIZ in MUSL
pbx_builtins: Add missing options documentation
res_pjsip_pubsub: update RLS to reflect the changes to the lists
res_agi: Fix xmldocs bug with set music.
res_config_pgsql: Add text-type column check in require_pgsql()
app_queue: Add QueueWithdrawCaller AMI action
ami: Improve substring parsing for disabled events.
xml.c, config,c: Add stylesheets and variable list string parsing
xmldoc: Fix issue with xmlstarlet validation
core: Config and XML tweaks needed for geolocation
Makefile: Allow XML documentation to exist outside source files
build: Refactor the earlier "basebranch" commit
jansson: Update bundled to 2.14 version.
func_channel: Add lastcontext and lastexten.
channel.c: Clean up debug level 1.
configs, LICENSE: remove pbx.digium.com.
documentation: Add since tag to xmldocs DTD
asterisk: Add macro for curl user agent.
res_stir_shaken: refactor utility function
app_voicemail: Emit warning if asking for nonexistent mailbox.
res_pjsip_pubsub: fix Batched Notifications stop working
res_pjsip_pubsub: provide a display name for RLS subscriptions
func_db: Add validity check for key names when writing.
cli: Add core dump info to core show settings.
documentation: Adds missing default attributes.
app_mp3: Document and warn about HTTPS incompatibility.
app_mf: Add max digits option to ReceiveMF.
ami: Allow events to be globally disabled.
taskprocessor.c: Prevent crash on graceful shutdown
app_queue: load queues and members from Realtime when needed
manager.c: Simplify AMI ModuleCheck handling
res_prometheus.c: missing module dependency
res_pjsip.c: Correct minor typos in 'realm' documentation.
manager.c: Generate valid XML if attribute names have leading digits.
build_tools/make_version: Fix bashism in comparison.
bundled_pjproject: Add additional multipart search utils
chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN
build: Add "basebranch" to .gitreview
res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup
cdr: allow disabling CDR by default on new channels
res_tonedetect: Fixes some logic issues and typos
func_frame_drop: Fix typo referencing wrong buffer
res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf
res_http_websocket: Add a client connection timeout
build: Rebuild configure and autoconfig.h.in
sched: fix and test a double deref on delete of an executing call back
app_queue.c: Queue don't play "thank-you" when here is no hold time announceme..
res_pjsip_sdp_rtp.c: Support keepalive for video streams.
build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD
main/utils: Implement ast_get_tid() for NetBSD
main: Enable rdtsc support on NetBSD
BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD
include: Remove unimplemented HMAC declarations
frame.h: Fix spelling typo
res_rtp_asterisk: Fix typo in flag test/set
bundled_pjproject: Fix srtp detection
res_pjsip: Make message_filter and session multipart aware
build: Fix issues building pjproject
res_pjsip: Add utils for checking media types
bundled_pjproject: Create generic pjsip_hdr_find functions
say.c: Prevent erroneous failures with 'say' family of functions.
documentation: Document built-in system and channel vars
pbx_variables: add missing ASTSBINDIR variable
bundled_pjproject: Make it easier to hack
utils.c: Remove all usages of ast_gethostbyname()
say.conf: fix 12pm noon logic
pjproject: Fix incorrect unescaping of tokens during parsing
app_queue.c: Support for Nordic syntax in announcements
dsp: Add define macro for DTMF_MATRIX_SIZE
ami: Add AMI event for Wink
cli: Add module refresh command
app_mp3: Throw warning on nonexistent stream
documentation: Add missing AMI documentation
tcptls.c: refactor client connection to be more robust
app_sf: Add full tech-agnostic SF support
app_queue: Fix hint updates, allow dup. hints
say.c: Honor requests for DTMF interruption.
res_pjsip_sdp_rtp: Preserve order of RTP codecs
bridge: Unlock channel during Local peer check.
test_time.c: Tolerate DST transitions
bundled_pjproject: Add more support for multipart bodies
ast_coredumper: Fix deleting results when output dir is set
pbx_variables: initialize uninitialized variable
app_queue.c: added DIALEDPEERNUMBER on outgoing channel
http.c: Add ability to create multiple HTTP servers
app.c: Throw warnings for nonexistent options
app_voicemail.c: Support for Danish syntax in VM
app_sendtext: Add ReceiveText application
strings: Fix enum names in comment examples
pbx_variables: Increase parsing capabilities of MSet
chan_sip: Fix crash when accessing RURI before initiating outgoing call
func_json: Adds JSON_DECODE function
configs: Updates to sample configs
pbx: Add variable substitution API for extensions
CHANGES: Correct reference to configuration file.
app_mf: Add full tech-agnostic MF support
xmldoc: Avoid whitespace around value for parameter/required.
progdocs: Fix Doxygen left-overs.
xmldoc: Correct definition for XML element 'matchInfo'.
progdocs: Update Makefile.
res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites.
channel: Short-circuit ast_channel_get_by_name() on empty arg.
res_rtp_asterisk: Addressing possible rtp range issues
apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL..
res: Fix for Doxygen.
res_fax_spandsp: Add spandsp 3.0.0+ compatibility
main: Fix for Doxygen.
progdocs: Fix for Doxygen, the hidden parts.
progdocs: Fix grouping for latest Doxygen.
documentation: Standardize examples
config.c: Prevent UB in ast_realtime_require_field.
app_voicemail: Refactor email generation functions
stir/shaken: Avoid a compiler extension of GCC.
progdocs: Remove outdated references in doxyref.h.
logger: use FUNCTION instead of PRETTY_FUNCTION
xmldoc: Fix for Doxygen.
astobj2.c: Fix core when ref_log enabled
channels: Fix for Doxygen.
bridge: Deny full Local channel pair in bridge.
res_tonedetect: Add call progress tone detection
rtp_engine: Add type field for JSON RTCP Report stasis messages
odbc: Fix for Doxygen.
parking: Fix for Doxygen.
res_ari: Fix for Doxygen.
frame: Fix for Doxygen.
ari-stubs: Avoid 'is' as comparism with an literal.
BuildSystem: Consistently allow 'ye' even for Jansson.
stasis: Fix for Doxygen.
app: Fix for Doxygen.
res_xmpp: Fix for Doxygen.
channel: Fix for Doxygen.
chan_iax2: Fix for Doxygen.
res_pjsip: Fix for Doxygen.
bridges: Fix for Doxygen.
addons: Fix for Doxygen.
apps: Fix for Doxygen.
tests: Fix for Doxygen.
progdocs: Avoid multiple use of section labels.
progdocs: Use Doxygen \example correctly.
bridge_channel: Fix for Doxygen.
progdocs: Avoid 'name' with Doxygen \file.
app_morsecode: Fix deadlock
app_read: Fix custom terminator functionality regression
res_pjsip_callerid: Fix OLI parsing
build_tools: Spelling fixes
contrib: Spelling fixes
codecs: Spelling fixes
formats: Spelling fixes
CREDITS: Spelling fixes
addons: Spelling fixes
configs: Spelling fixes
doc: Spelling fixes
menuselect: Spelling fixes
include: Spelling fixes
UPGRADE.txt: Spelling fixes
bridges: Spelling fixes
apps: Spelling fixes
channels: Spelling fixes
tests: Spelling fixes
CHANGES: Spelling fixes
funcs: Spelling fixes
pbx: Spelling fixes
main: Spelling fixes
utils: Spelling fixes
Makefile: Spelling fixes
res: Spelling fixes
rest-api-templates: Spelling fixes
agi: Spelling fixes
CI: Rename 'master' node to 'built-in'
BuildSystem: In POSIX sh, == in place of = is undefined.
pbx.c: Don't remove dashes from hints on reload.
sig_analog: Fix truncated buffer copy
app_voicemail: Fix phantom voicemail bug on rerecord
chan_iax2: Allow both secret and outkey at dial time
res_snmp: As build tool, prefer pkg-config over net-snmp-config.
res_config_sqlite: Remove deprecated module.
stasis: Avoid 'dispatched' as unused variable in normal mode.
various: Fix GCC 11.2 compilation issues.
ast_coredumper: Refactor to better find things
strings/json: Add string delimter match, and object create with vars methods
STIR/SHAKEN: Option split and response codes.
app_queue: Add LoginTime field for member in a queue.
res_speech: Add a type conversion, and new engine unregister methods
various: Fix GCC 11 compilation issues.
apps/app_playback.c: Add 'mix' option to app_playback
BuildSystem: Check for alternate openssl packages
func_talkdetect.c: Fix logical errors in silence detection.
configure: Remove unused OpenSSL SRTP check.
build: prevent binary downloads for non x86 architectures
main/stun.c: fix crash upon STUN request timeout
Makefile: Use basename in a POSIX-compliant way.
pbx_ael: Fix crash and lockup issue regarding 'ael reload'
chan_iax2: Add encryption for RSA authentication
res_pjsip_t38: bind UDPTL sessions like RTP
app_read: Fix null pointer crash
res_rtp_asterisk: fix memory leak
main/say.c: Support future dates with Q and q format params
res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts
ari: Ignore invisible bridges when listing bridges.
func_vmcount: Add support for multiple mailboxes
message.c: Support 'To' header override with AMI's MessageSend.
func_channel: Add CHANNEL_EXISTS function.
app_queue: Fix hint updates for included contexts
res_http_media_cache.c: Compare unaltered MIME types.
logger: Add custom logging capabilities
app_externalivr.c: Fix mixed leading whitespace in source code.
res_rtp_asterisk.c: Fix build failure when not building with pjproject.
pjproject: Add patch to fix trailing whitespace issue in rtpmap
app_mp3: Force output to 16 bits in mpg123
res_pjsip_caller_id: Add ANI2/OLI parsing
app_mf: Add channel agnostic MF sender
app_stack: Include current location if branch fails
test_http_media_cache.c: Fix copy/paste error during test deregistration.
resource_channels.c: Fix external media data option
func_strings: Add STRBETWEEN function
test_abstract_jb.c: Fix put and put_out_of_order memory leaks.
func_env: Add DIRNAME and BASENAME functions
func_sayfiles: Retrieve say file names
res_tonedetect: Tone detection module
res_snmp: Add -fPIC to _ASTCFLAGS
app_voicemail.c: Ability to silence instructions if greeting is present.
term.c: Add support for extended number format terminfo files.
res_srtp: Disable parsing of not enabled cryptos
dns.c: Load IPv6 DNS resolvers if configured.
bridge_softmix: Suppress error on topology change failure
resource_channels.c: Fix wrong external media parameter parse
config_options: Handle ACO arrays correctly in generated XML docs.
chan_iax2: Add ANI2/OLI information element
pbx_ael: Fix crash and lockup issue regarding 'ael reload'
app_read: Allow reading # as a digit
res_rtp_asterisk: Automatically refresh stunaddr from DNS
bridge_basic: Change warning to verbose if transfer cancelled
app_queue: Don't reset queue stats on reload
res_rtp_asterisk: sqrt(.) requires the header math.h.
dialplan: Add one static and fix two whitespace errors.
sig_analog: Changes to improve electromechanical signalling compatibility
media_cache: Don't lock when curl the remote file
res_pjproject: Allow mapping to Asterisk TRACE level
app_milliwatt: Timing fix
func_math: Return integer instead of float if possible
app_morsecode: Add American Morse code
func_scramble: Audio scrambler function
app_originate: Add ability to set codecs
BuildSystem: Remove two dead exceptions for compiler Clang.
chan_alsa, chan_sip: Add replacement to moduleinfo
res_monitor: Disable building by default.
muted: Remove deprecated application.
conf2ael: Remove deprecated application.
res_config_sqlite: Remove deprecated module.
chan_vpb: Remove deprecated module.
chan_misdn: Remove deprecated module.
chan_nbs: Remove deprecated module.
chan_phone: Remove deprecated module.
chan_oss: Remove deprecated module.
cdr_syslog: Remove deprecated module.
app_dahdiras: Remove deprecated module.
app_nbscat: Remove deprecated module.
app_image: Remove deprecated module.
app_url: Remove deprecated module.
app_fax: Remove deprecated module.
app_ices: Remove deprecated module.
app_mysql: Remove deprecated module.
cdr_mysql: Remove deprecated module.
mgcp: Remove dead debug code
policy: Deprecate modules and add versions to others.
func_frame_drop: New function
aelparse: Accept an included context with timings.
format_ogg_speex: Implement a "not supported" write handler
cdr_adaptive_odbc: Prevent filter warnings
app_queue: Allow streaming multiple announcement files
res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern
res_statsd: handle non-standard meter type safely
app_dtmfstore: New application to store digits
codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother
res_http_media_cache: Cleanup audio format lookup in HTTP requests
docs: Remove embedded macro in WaitForCond XML documentation.
Update AMI and ARI versions for Asterisk 20.
AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
AST-2021-008 - chan_iax2: remote crash on unsupported media format
AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.
res_stasis_playback: Check for chan hangup on play_on_channels
res_http_media_cache.c: Fix merge errors from 18 -> master
res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.
res_http_media_cache.c: Parse media URLs to find extensions.
main/cdr.c: Correct Party A selection.
stun: Emit warning message when STUN request times out
app_reload: New Reload application
res_ari: Fix audiosocket segfault
res_pjsip_config_wizard.c: Add port matching support.
app_waitforcond: New application
res_stasis_playback: Send PlaybackFinish event only once for errors
jitterbuffer: Correct signed/unsigned mismatch causing assert
app_dial: Expanded A option to add caller announcement
core: Don't play silence for Busy() and Congestion() applications.
res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress
res_pjsip_messaging: Overwrite user in existing contact URI
res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter
pbx_builtins: Corrects SayNumber warning
func_lock: Add "dialplan locks show" cli command.
func_lock: Prevent module unloading in-use module.
func_lock: Fix memory corruption during unload.
func_lock: Fix requesters counter in error paths.
app_originate: Allow setting Caller ID and variables
menuselect: Fix description of several modules.
app_confbridge: New ConfKick() application
res_pjsip_dtmf_info: Hook flash
app_confbridge: New option to prevent answer supervision
sip_to_pjsip: Fix missing cases
res_pjsip_messaging: Refactor outgoing URI processing
func_math: Three new dialplan functions
STIR/SHAKEN: Add Date header, dest->tn, and URL checking.
res_pjsip: On partial transport reload also move factories.
func_volume: Add read capability to function.
stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing
res_pjsip.c: Support endpoints with domain info in username
res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.
asterisk: We've moved to Libera Chat!
res_rtp_asterisk: make it possible to remove SOFTWARE attribute
res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs
res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml
AMI: Add AMI event to expose hook flash events
app_voicemail: Configurable voicemail beep
main/file.c: Don't throw error on flash event.
chan_sip: Expand hook flash recognition.
pjsip: Add patch for resolving STUN packet lifetime issues.
chan_pjsip: Correct misleading trace message
STIR/SHAKEN: Switch to base64 URL encoding.
STIR/SHAKEN: OPENSSL_free serial hex from openssl.
STIR/SHAKEN: Fix certificate type and storage.
translate.c: Avoid refleak when checking for a translation path
res_rtp_asterisk: More robust timestamp checking
chan_local: Skip filtering audio formats on removed streams.
res_pjsip.c: OPTIONS processing can now optionally skip authentication
translate.c: Take sampling rate into account when checking codec's buffer size
svn: Switch to https scheme.
res_pjsip: Update documentation for the auth object
res_aeap: Add basic config skeleton and CLI commands.
bridge_channel_write_frame: Check for NULL channel
loader.c: Speed up deprecation metadata lookup
res_prometheus: Clone containers before iterating
loader: Output warnings for deprecated modules.
res_rtp_asterisk: Fix standard deviation calculation
res_rtp_asterisk: Don't count 0 as a minimum lost packets
res_rtp_asterisk: Statically declare rtp_drop_packets_data object
res_rtp_asterisk: Only raise flash control frame on end.
res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
res_pjsip: Give error when TLS transport configured but not supported.
time: Add timeval create and unit conversion functions
app_queue: Add alembic migration to add ringinuse to queue_members.
modules.conf: Fix more differing usages of assignment operators.
logger.conf.sample: Add more debug documentation.
logging: Add .log to samples and update asterisk.logrotate.
app_queue.c: Remove dead 'updatecdr' code.
queues.conf.sample: Correct 'context' documentation.
logger: Console sessions will now respect logger.conf dateformat= option
app_queue.c: Don't crash when realtime queue name is empty.
res_pjsip_session: Make reschedule_reinvite check for NULL topologies
app_queue: Only send QueueMemberStatus if status changes.
core_unreal: Fix deadlock with T.38 control frames.
res_pjsip: Add support for partial transport reload.
menuselect: exit non-zero in case of failure on --enable|disable options.
res_rtp_asterisk: Force resync on SSRC change.
menuselect: Add ability to set deprecated and removed versions.
xml: Allow deprecated_in and removed_in for MODULEINFO.
xml: Embed module information into core XML documentation.
documentation: Fix non-matching module support levels.
channel: Fix crash in suppress API.
func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds
app.h: Fix -Werror=zero-length-bounds compile errors in dev mode.
app_dial.c: Only send DTMF on first progress event.
res_format_attr_*: Parameter Names are Case-Insensitive.
chan_iax2: System Header strings is included via asterisk.h/compat.h.
modules.conf: Fix differing usage of assignment operators.
strings.h: ast_str_to_upper() and _to_lower() are not pure.
res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
res/res_rtp_asterisk: generate new SSRC on native bridge end
sorcery: Add support for more intelligent reloading.
res_pjsip_refer: Move the progress dlg release to a serializer
res_pjsip_registrar: Include source IP and port in log messages.
asterisk: Update copyright.
AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
res_format_attr_h263: Generate valid SDP fmtp for H.263+.
res_pjsip_nat: Don't rewrite Contact on REGISTER responses.
channel: Fix memory leak in suppress API.
res_rtp_asterisk: Check remote ICE reset and reset local ice attrb
pjsip: Generate progress (once) when receiving a 180 with a SDP
main: With Dutch language year after 2020 is not spoken in say.c
res_pjsip: dont return early from registration if init auth fails
res_fax: validate the remote/local Station ID for UTF-8 format
app_page.c: Don't fail to Page if beep sound file is missing
res_pjsip_refer: Refactor progress locking and serialization
res_rtp_asterisk: Add packet subtype during RTCP debug when relevant
res_pjsip_session: Always produce offer on re-INVITE without SDP.
res_odbc_transaction: correctly initialise forcecommit value from DSN.
res_pjsip_session.c: Check topology on re-invite.
res_config_pgsql: Limit realtime_pgsql() to return one (no more) record.
app_queue: Fix conversion of complex extension states into device states
app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS
chan_sip: Filter pass-through audio/video formats away, again.
func_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute.
AST-2021-002: Remote crash possible when negotiating T.38
rtp: Enable srtp replay protection
res_pjsip_diversion: Fix adding more than one histinfo to Supported
res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
pjsip: Make modify_local_offer2 tolerate previous failed SDP.
res_pjsip_refer: Always serialize calls to refer_progress_notify
core_unreal: Fix T.38 faxing when using local channels.
format_wav: Support of MIME-type for wav16
chan_sip: Allow [peer] without audio (text+video).
chan_iax2.c: Require secret and auth method if encryption is enabled
app_read: Release tone zone reference on early return.
chan_sip: Set up calls without audio (text+video), again.
chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable
channel: Set up calls without audio (text+video), again.
res/res_pjsip.c: allow user=phone when number contain *#
chan_sip: SDP: Reject audio streams correctly.
main/frame: Add missing control frame names to ast_frame_subclass2str
res_musiconhold: Add support of various URL-schemes by MoH.
AC_HEADER_STDC causes a compile failure with autoconf 2.70
pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.
res_pjsip_session: Avoid sometimes-uninitialized warning with Clang.
res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet
chan_pjsip.c: Add parameters to frame in indicate.
res/res_pjsip_session.c: Check that media type matches in function ast_sip_ses..
Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
chan_sip: SDP: Sidestep stream parsing when its media is disabled.
chan_pjsip: Assign SIPDOMAIN after creating a channel
chan_pjsip: Stop queueing control frames twice on outgoing channels
contrib/systemd: Added note on common issues with systemd and asterisk
Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other st..
func_lock: fix multiple-channel-grant problems.
pbx_lua: Add LUA_VERSIONS environment variable to ./configure.
app_mixmonitor: cleanup datastore when monitor thread fails to launch
app_voicemail: Prevent deadlocks when out of ODBC database connections
chan_pjsip: Incorporate channel reference count into transfer_refer().
pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type
asterisk: Export additional manager functions
res_pjsip: Prevent segfault in UDP registration with flow transports
codecs: Remove test-law.
res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
chan_vpb.cc: Fix compile errors.
res_pjsip_session.c: Fix compiler warnings.
res_pjsip_session: Fixed NULL active media topology handle
app_chanspy: Spyee information missing in ChanSpyStop AMI Event
res_ari: Fix wrong media uri handle for channel play
logger.c: Automatically add a newline to formats that don't have one
res_pjsip_nat.c: Create deep copies of strings when appropriate
res_musiconhold: Don't crash when real-time doesn't return any entries
res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
pjsip: Match lifetime of INVITE session to our session.
res_http_media_cache.c: Set reasonable number of redirects
Introduce astcachedir, to be used for temporary bucket files
media_cache: Fix reference leak with bucket file metadata
res_pjsip_stir_shaken: Fix module description
voicemail: add option 'e' to play greetings as early media
loader: Sync load- and build-time deps.
CHANGES: Remove already applied CHANGES update
res_pjsip: set Accept-Encoding to identity in OPTIONS response
chan_sip: Remove unused sip_socket->port.
bridge_basic: Fixed setup of recall channels
modules.conf: Align the comments for more conclusiveness.
app_queue: Fix deadlock between update and show queues
res_pjsip_outbound_registration.c: Use our own scheduler and other stuff
pjsip_scheduler.c: Add type ONESHOT and enhance cli show command
sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data
res_pjsip/config_transport: Load and run without OpenSSL.
res_stir_shaken: Include OpenSSL headers where used actually.
func_curl.c: Allow user to set what return codes constitute a failure.
AST-2020-001 - res_pjsip: Return dialog locked and referenced
AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
sip_to_pjsip.py: Handle #include globs and other fixes
Compiler fixes for GCC with -Og
Compiler fixes for GCC when printf %s is NULL
Compiler fixes for GCC with -Os
chan_sip: On authentication, pick MD5 for sure.
main/say: Work around gcc 9 format-truncation false positive
res_pjsip, res_pjsip_session: initialize local variables
install_prereq: Add GMime 3.0.
BuildSystem: Enable Lua 5.4.
res_pjsip_session: Restore calls to ast_sip_message_apply_transport()
features.conf.sample: Sample sound files incorrectly quoted
logger.conf.sample: add missing comment mark
res_pjsip: Adjust outgoing offer call pref.
tcptls.c: Don't close TCP client file descriptors more than once
resource_endpoints.c: memory leak when providing a 404 response
Logging: Add debug logging categories
pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
app_confbridge/bridge_softmix: Add ability to force estimated bitrate
app_voicemail.c: Document VMSayName interruption behavior
res_pjsip_sdp_rtp: Fix accidentally native bridging calls
res_musiconhold: Load all realtime entries, not just the first
channels: Don't dereference NULL pointer
res_pjsip_diversion: fix double 181
res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs
dsp.c: Update calls to ast_format_cmp to check result properly
res_pjsip_session: Fix stream name memory leak.
func_curl.c: Prevent crash when using CURLOPT(httpheader)
res_musiconhold: Start playlist after initial announcement
res_pjsip_session: Fix session reference leak.
res_stasis.c: Add compare function for bridges moh container
logger.h: Fix ast_trace to respect scope_level
chan_sip.c: Don't build by default
audiosocket: Fix module menuselect descriptions
bridge_softmix/sfu_topologies_on_join: Ignore topology change failures
res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined
res_pjsip_diversion: implement support for History-Info
format_cap: Perform codec lookups by pointer instead of name
res_pjsip_session: Fix issue with COLP and 491
debugging: Add enough to choke a mule
res_pjsip_session: Handle multi-stream re-invites better
realtime: Increased reg_server character size
res_stasis.c: Added video_single option for bridge creation
Bridging: Use a ref to bridge_channel's channel to prevent crash.
res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a..
conversions: Add string to signed integer conversion functions
app_queue: Fix leave-empty not recording a call as abandoned
ast_coredumper: Fix issues with naming
parking: Copy parker UUID as well.
sip_nat_settings: Update script for latest Linux.
samples: Fix keep_alive_interval default in pjsip.conf.
chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution
pbx: Fix hints deadlock between reload and ExtensionState.
logger.c: Added a new log formatter called "plain"
res_speech: Bump reference on format object
res_pjsip_diversion: handle 181
app_voicemail: Process urgent messages with mailcmd
app_queue: Member lastpause time reseting
res_pjsip_session: Don't aggressively terminate on failed re-INVITE.
bridge_channel: Ensure text messages are zero terminated
res_musiconhold.c: Use ast_file_read_dir to scan MoH directory
scope_trace: Added debug messages and added additional macros
stream.c: Added 2 more debugging utils and added pos to stream string
chan_sip: Clear ToHost property on peer when changing to dynamic host
ACN: Changes specific to the core
Makefile: Fix certified version numbers
res_musiconhold.c: Prevent crash with realtime MoH
res_pjsip: Fix codec preference defaults.
vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
pjproject: clone sdp to protect against (nat) modifications
utils.c: NULL terminate ast_base64decode_string.
ACN: Configuration renaming for pjsip endpoint
res_stir_shaken: Fix memory allocation error in curl.c
res_pjsip_session: Ensure reused streams have correct bundle group
res_pjsip_registrar: Don't specify an expiration for static contacts.
utf8.c: Add UTF-8 validation and utility functions
stasis_bridge.c: Fixed wrong video_mode shown
vector.h: Add AST_VECTOR_SORT()
CI: Force publishAsteriskDocs to use python2
websocket / pjsip: Increase maximum packet size.
Prepare master for the next Asterisk version
acl.c: Coerce a NULL pointer into the empty string
pjsip: Include timer patch to prevent cancelling timer 0.

User Notes:

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.
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.
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 cha..

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.
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@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.
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 par..

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

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.

Upgrade Notes:

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.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 cha..

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.
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.
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
#37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
#39: [Bug]: Remove .gitreview from repository.
#43: [Bug]: Link to trademark policy is no longer correct
#45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
#46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
#48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
#52: [improvement]: Add local optimization begin cel event
#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
#71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
#73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
#84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
#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
#96: [bug]: make install-logrotate causes logrotate to fail on service restart
#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
#118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
#120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
#122: [new-feature]: res_musiconhold: Add looplast option
#129: [bug]: res_speech_aeap: Crash due to NULL format on setup
#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
#170: [improvement]: app_voicemail - add CLI commands to manipulate messages
#174: [bug]: app_voicemail imap compile errors
#179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
#181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
#193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
#200: [bug]: Regression: In app.h an enum is used before its declaration.
#202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
#205: [new-feature]: sig_analog: Allow flash to time out to silent hold
#224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
#226: [improvement]: Apply contact_user to incoming calls
#230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
#233: [bug]: Deadlock with MixMonitorMute AMI action
#240: [new-feature]: sig_analog: Add Called Subscriber Held capability
#242: [new-feature]: logger: Allow filtering logs in CLI by channel
#248: [bug]: core_local: Local channels cannot have slashes in the destination
#253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
#255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
#260: [bug]: maxptime must be changed to multiples of 20
#263: [bug]: download_externals doesn't always handle versions correctly
#265: [bug]: app_macro isn't locking around channel datastore access
#267: [bug]: ari: refer with display_name key in request body leads to crash
#274: [bug]: Syntax Error in SQL Code
#275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
#277: [bug]: pbx.c: Compiler error with gcc 12.2
#281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
#286: [improvement]: chan_iax2: Improve authentication debugging
#289: [new-feature]: Add support for deleting channel and global variables
#294: [improvement]: chan_dahdi: Improve call pickup documentation
#298: [improvement]: chan_rtp: Implement RTP glue
#301: [bug]: Number of ICE TURN threads continually growing
#303: [bug]: SpeechBackground never exits
#308: [bug]: chan_console: Deadlock when hanging up console channels
#315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/
#316: [bug]: Privilege Escalation in Astrisk's Group permissions.
#319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
#321: [bug]: Performance suffers unnecessarily when debugging deadlocks
#325: [bug]: hangup after beep to avoid waiting for timeout
#330: [improvement]: Add cel user event helper function
#337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
#341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
#345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
#349: [improvement]: Add libjwt to third-party
#351: [improvement]: Refactor res_stir_shaken to use libjwt
#352: [bug]: Update qualify_timeout documentation to include DNS note
#354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
#356: [new-feature]: app_directory: Add ADSI support.
#360: [improvement]: Update documentation for CHANGES/UPGRADE files
#362: [improvement]: Speed up ARI command processing
#379: [bug]: Orphaned taskprocessors cause shutdown delays
#384: [bug]: Unnecessary re-INVITE after answer
#388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
#396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
#398: [new-feature]: app_voicemail: Add AMI event for password change
#406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
#409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
#423: [improvement]: func_lock: Add missing see-also refs
#425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
#428: [bug]: cli: Output is truncated from "config show help"
#430: [bug]: Fix broken links
#440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
#442: [bug]: func_channel: Some channel options are not settable
#445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
#458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
#462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
#465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
#482: [improvement]: manager.c: Improve clarity of "manager show connected" output
#492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
#500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
#503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
#505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
#509: [bug]: res_pjsip: Crash when looking up transport state in use
#513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
#520: [improvement]: menuselect: Use more specific error message.
#527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
#529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
#530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
#533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
#539: [bug]: Existence of logger.xml causes linking failure
#551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
#560: [bug]: EndIf() causes next priority to be skipped
#565: [bug]: Application Read() returns immediately
#569: [improvement]: Add option to interleave input and output frames on spied channel
#572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
#582: [improvement]: Reduce unneeded logging during startup and shutdown
#586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
#588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
#592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
#595: [improvement]: dsp.c: Fix and improve confusing warning message.
#597: [bug]: wrong MOS calculation
#601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
#619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
#629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
#634: [bug]: make install doesn't create the stir_shaken cache directory
#636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
#645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c

An additional 751 ASTERISK-* issues were closed.

asterisk/asterisk certified-20.7-cert1-rc1 Asterisk Release certified-20.7-cert1-rc1 on GitHub

Change Log for Release asterisk-certified-20.7-cert1-rc1

Links:

Summary:

User Notes:

app_dial: Add dial time for progress/ringing.

app_voicemail: Allow preventing mark messages as urgent.

Stir/Shaken Refactor

Upgrade bundled pjproject to 2.14.

app_speech_utils.c: Allow partial speech results.

app_chanspy: Add 'D' option for dual-channel audio

chan_dahdi: Allow MWI to be manually toggled on channels.

app_dial: Add option "j" to preserve initial stream topology of caller

logger: Add channel-based filtering.

chan_pjsip: Add PJSIPHangup dialplan app and manager action

app_voicemail: Add AMI event for mailbox PIN changes.

res_speech: allow speech to translate input channel

res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..

res_stasis: signal when new command is queued

lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS

file.c: Add ability to search custom dir for sounds

make_buildopts_h, et. al. Allow adding all cflags to buildopts.h

chan_rtp: Implement RTP glue for UnicastRTP channels

app_queue: periodic announcement configurable start time.

variables: Add additional variable dialplan functions.

sig_analog: Add Called Subscriber Held capability.

res_pjsip_header_funcs: Make prefix argument optional.

core/ari/pjsip: Add refer mechanism

chan_dahdi: Allow autoreoriginating after hangup.

sig_analog: Allow three-way flash to time out to silence.

res_pjsip: Enable TLS v1.3 if present.

app_queue: Add support for applying caller priority change immediately.

Adds manager actions to allow move/remove/forward individual messages in a par..

app_voicemail: add CLI commands for message manipulation

sig_analog: Allow immediate fake ring to be suppressed.

AMI: Add parking position parameter to Park action

res_musiconhold: Add option to loop last file.

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

cel: add local optimization begin event

chan_dahdi: Add dialmode option for FXS lines.

Upgrade Notes:

Stir/Shaken Refactor

app.c: Allow ampersands in playback lists to be escaped.

pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.

res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..

app_queue: Preserve reason for realtime queues

cel: add local optimization begin event

Closed Issues:

asterisk/asterisk certified-20.7-cert1-rc1
Asterisk Release certified-20.7-cert1-rc1

on GitHub