Log message:
Update to Asterisk 21.12.0.
## Change Log for Release asterisk-21.12.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
### Summary:
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
### User Notes:
- #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
  Added a new option to HANGUPCAUSE to access additional
  information about hangup reason. Reason headers from pjsip
  could be read using 'tech_extended' cause type.
- #### chan_dahdi: Add DAHDI_CHANNEL function.
  The DAHDI_CHANNEL function allows for getting/setting
  certain properties about DAHDI channels from the dialplan.
### Upgrade Notes:
- #### res_audiosocket: add message types for all slin sample rates
  New audiosocket message types 0x11 - 0x18 has been added
  for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
  slin192 audio. External applications using audiosocket may need to be
  updated to support these message types if the audiosocket channel is
  created with one of these audio formats.
## Issue and Commit Detail:
### Closed Issues:
  - 1340: [bug]: comfort noise packet corrupted
  - 1419: [bug]: static code analysis issues in app_adsiprog.c
  - 1422: [bug]: static code analysis issues in apps/app_externalivr.c
  - 1425: [bug]: static code analysis issues in apps/app_queue.c
  - 1434: [improvement]: pbx_variables: Create real channel for dialplan eval \ 
CLI command
  - 1436: [improvement]: res_cliexec: Avoid unnecessary cast to char*
  - 1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function
  - 1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with \ 
PJSIP_TRANSFER_HANDLING(ari-only)
  - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock \ 
(`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup
  - 1525: [bug]: chan_websocket: fix use of raw payload variable for string \ 
comparison in process_text_message
  - 1539: [bug]: safe_asterisk without TTY doesn't log to file
  - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361
  - 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map \ 
channel backend

Log message:
Upgrade to Asterisk 21.11.0.
## Change Log for Release asterisk-21.11.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
### Summary:
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
### User Notes:
- #### app_queue.c: Add new global 'log_unpause_on_reason_change'
  Add new global option 'log_unpause_on_reason_change' that
  is default disabled. When enabled cause addition of UNPAUSE event on
  every re-PAUSE with reason changed.
- #### pbx_builtins: Allow custom tone for WaitExten.
  The tone used while waiting for digits in WaitExten
  can now be overridden by specifying an argument for the 'd'
  option.
- #### res_tonedetect: Add option for TONE_DETECT detection to auto stop.
  The 'e' option for TONE_DETECT now allows detection to
  be disabled automatically once the desired number of matches have
  been fulfilled, which can help prevent race conditions in the
  dialplan, since TONE_DETECT does not need to be disabled after
  a hit.
- #### sorcery: Prevent duplicate objects and ensure missing objects are created \ 
on u..
  Users relying on Sorcery multiple writable backends configurations
  (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes
  in sorcery.conf to ensure missing objects are recreated after temporary backend
  failures. Default behavior remains unchanged unless explicitly enabled.
- #### chan_websocket: Allow additional URI parameters to be added to the \ 
outgoing URI.
  A new WebSocket channel driver option `v` has been added to the
  Dial application that allows you to specify additional URI parameters on
  outgoing connections. Run `core show application Dial` from the Asterisk CLI
  to see how to use it.
- #### app_chanspy: Add option to not automatically answer channel.
  ChanSpy and ExtenSpy can now be configured to not
  automatically answer the channel by using the 'N' option.
- #### cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
  Enabling the tracking of the
  STREAM_BEGIN and the STREAM_END event
  types in cel.conf will log media files and
  music on hold played to each channel.
  The STREAM_BEGIN event's extra field will
  contain a JSON with the file details (path,
  format and language), or the class name, in
  case of music on hold is played. The DTMF
  event's extra field will contain a JSON with
  the digit and the duration in milliseconds.
- #### res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
  Options are now available in the menuselect "Resource Modules"
  category that allow you to enable the AES_192, AES_256 and AES_GCM
  cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support
  them but modern versions do.  Previously, the only way to enable them was
  to set the CFLAGS environment variable when running ./configure.
  The default setting is to disable them preserving existing behavior.
- #### cdr: add CANCEL dispostion in CDR
  A new CDR option "canceldispositionenabled" has been added
  that when set to true, the NO ANSWER disposition will be split into
  two dispositions: CANCEL and NO ANSWER. The default value is 'no'
- #### func_curl: Allow auth methods to be set.
  The httpauth field in CURLOPT now allows the authentication
  methods to be set.
- #### Media over Websocket Channel Driver
  A new channel driver "chan_websocket" is now available. It can
  exchange media over both inbound and outbound websockets and will both frame
  and re-time the media it receives.
  See https://s.asterisk.net/mow for more information.
  The ARI channels/externalMedia API now includes support for the
### Upgrade Notes:
### Developer Notes:
- #### ARI: Add command to indicate progress to a channel
  A new ARI endpoint is available at `/channels/{channelId}/progress` to \ 
indicate progress to a channel.
- #### options:  Change ast_options from ast_flags to ast_flags64.
  The 32-bit ast_options has no room left to accomodate new
  options and so has been converted to an ast_flags64 structure. All internal
  references to ast_options have been updated to use the 64-bit flag
  manipulation macros.  External module references to the 32-bit ast_options
  should continue to work on little-endian systems because the
  least-significant bytes of a 64 bit integer will be in the same location as a
  32-bit integer.  Because that's not the case on big-endian systems, we've
  swapped the bytes in the flags manupulation macros on big-endian systems
  so external modules should still work however you are encouraged to test.
## Issue and Commit Detail:
### Closed Issues:
  - 401: [bug]: app_dial: Answer Gosub option passthrough regression
  - 927: [bug]: no audio when media source changed during the call
  - 1176: [bug]: ast_slinear_saturated_multiply_float produces potentially \ 
audible distortion artifacts
  - 1259: [bug]: New TenantID feature doesn't seem to set CDR for incoming calls
  - 1260: [bug]: Asterisk sends RTP audio stream before ICE/DTLS completes
  - 1269: [bug]: MixMonitor with D option produces corrupt file
  - 1273: [bug]: When executed with GotoIf, the action Redirect does not take \ 
effect and causes confusion in dialplan execution.
  - 1280: [improvement]: logging playback of audio per channel
  - 1289: [bug]: sorcery - duplicate objects from multiple backends and backend \ 
divergence on update
  - 1301: [bug]: sig_analog: fgccamamf doesn't handle STP, STP2, or STP3
  - 1304: [bug]: FLUSH_MEDIA does not reset frame_queue_length in WebSocket channel
  - 1305: [bug]: Realtime incorrectly falls back to next backend on \ 
record-not-found (SQL_NO_DATA), causing incorrect behavior and delay
  - 1307: [improvement]: ast_tls_cert: Allow certificate validity to be configurable
  - 1309: [bug]: Crash with C++ alternative storage backend enabled
  - 1315:  [bug]: When executed with dialplan, the action Redirect does not take \ 
effect.
  - 1317: [bug]: AGI command buffer overflow with long variables
  - 1321: [improvement]: app_agent_pool: Remove obsolete documentation
  - 1323: [new-feature]: add CANCEL dispostion in CDR
  - 1327: [bug]: res_stasis_device_state: can't delete ARI Devicestate after \ 
asterisk restart
  - 1332: [new-feature]: func_curl: Allow auth methods to be set
  - 1349: [bug]: Race condition on redirect can cause missing Diversion header
  - 1352: [improvement]: Websocket channel with custom URI
  - 1353: [bug]: AST_DATA_DIR/sounds/custom directory not searched
  - 1358: [new-feature]: app_chanspy: Add option to not automatically answer channel
  - 1364: [bug]: bridge.c: BRIDGE_NOANSWER not always obeyed
  - 1366: [improvement]: func_frame_drop: Handle allocation failure properly
  - 1369: [bug]: test_res_prometheus: Compilation failure in devmode due to \ 
curlopts not using long type
  - 1371: [improvement]: func_frame_drop: Add debug messages for frames that can \ 
be dropped
  - 1375: [improvement]: dsp.c: Improve logging in tone_detect().
  - 1378: [bug]: chan_dahdi: dialmode feature is not properly reset between calls
  - 1380: [bug]: sig_analog: Segfault due to calling strcmp on NULL
  - 1384: [bug]: chan_websocket: asterisk crashes on hangup after \ 
STOP_MEDIA_BUFFERING command with id
  - 1386: [bug]: enabling announceposition_only_up prevents any queue position \ 
announcements
  - 1390: [improvement]: res_tonedetect: Add option to automatically end \ 
detection in TONE_DETECT
  - 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
  - 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable
  - 1401: [bug]: app_waitfornoise timeout is always less then configured because \ 
of time() usage
  - 1457: [bug]: segmentation fault because of a wrong ari config
  - 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
  - 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk \ 
change to stop media flow before DTLS completes
### Commit List:
-  res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
-  chan_websocket: Fix codec validation and add passthrough option.
-  res_ari: Ensure outbound websocket config has a websocket_client_id.
-  chan_websocket.c: Add DTMF messages
-  app_queue.c: Add new global 'log_unpause_on_reason_change'
-  app_waitforsilence.c: Use milliseconds to calculate timeout time
-  Fix missing ast_test_flag64 in extconf.c
-  pbx_builtins: Allow custom tone for WaitExten.
-  res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-  app_queue: fix comparison for announce-position-only-up
-  sig_analog: Skip Caller ID spill if usecallerid=no.
-  chan_dahdi: Fix erroneously persistent dialmode.
-  chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
-  sig_analog: Fix SEGV due to calling strcmp on NULL.
-  ARI: Add command to indicate progress to a channel
-  dsp.c: Improve debug logging in tone_detect().
-  res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
-  app_chanspy: Add option to not automatically answer channel.
-  xmldoc.c: Fix rendering of CLI output.
-  func_frame_drop: Add debug messages for dropped frames.
-  test_res_prometheus: Fix compilation failure on Debian 13.
-  func_frame_drop: Handle allocation failure properly.
-  pbx_lua.c: segfault when pass null data to term_color function
-  bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel.
-  res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
-  app_dial.c: Moved channel lock to prevent deadlock
-  file.c: with "sounds_search_custom_dir = yes", search \ 
"custom" directory
-  cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-  channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.
-  res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-  cdr: add CANCEL dispostion in CDR
-  func_curl: Allow auth methods to be set.
-  options:  Change ast_options from ast_flags to ast_flags64.
-  res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA)
-  app_agent_pool: Remove documentation for removed option.
-  res_agi: Increase AGI command buffer size from 2K to 8K
-  ast_tls_cert: Make certificate validity configurable.
-  cdr.c: Set tenantid from party_a->base instead of chan->base.
-  app_mixmonitor:  Update the documentation concerning the "D" option.
-  sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf.
-  chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
-  chan_pjsip.c: Change SSRC after media source change
-  Media over Websocket Channel Driver
-  bundled_pjproject: Avoid deadlock between transport and transaction
-  utils.h: Add rounding to float conversion to int.
-  res_musiconhold.c: Ensure we're always locked around music state access.
-  res_musiconhold.c: Annotate when the channel is locked.
-  res_musiconhold: Appropriately lock channel during start.
## Change Log for Release asterisk-21.10.2
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
### Summary:
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - \ 
[GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j): \ 
A specifically malformed Authorization header in an incoming SIP request can \ 
cause Asterisk to crash
### Commit Authors:
- George Joseph: (1)
## Issue and Commit Detail:
### Closed Issues:
  - !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an \ 
incoming SIP request can cause Asterisk to crash
### Commit List:
-  res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
### Commit Details:
#### res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
  Author: George Joseph
  Date:   2025-08-28
  In the highly-unlikely event that get_authorization_hdr() couldn't find an
  Authorization header in a request, trying to get the digest algorithm
  would cauase a SEGV.  We now check that we have an auth header that matches
  the realm before trying to get the algorithm from it.
  Resolves: #GHSA-64qc-9x89-rx5j

Log message:
*: rev bump for curl

Log message:
*: recursive bump for tiff growing lerc dependency

Log message:
Update to Asterisk 21.10.1.  This is a security update.
## Change Log for Release asterisk-21.10.1
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.1.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.0...21.10.1)
### Summary:
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
  - \ 
[GHSA-mrq5-74j5-f5cr](https://github.com/asterisk/asterisk/security/advisories/GHSA-mrq5-74j5-f5cr): \ 
Remote DoS and possible RCE in asterisk/res/res_stir_shaken/verification.c
  - \ 
[GHSA-v9q8-9j8m-5xwp](https://github.com/asterisk/asterisk/security/advisories/GHSA-v9q8-9j8m-5xwp): \ 
Uncontrolled Search-Path Element in safe_asterisk script may allow local \ 
privilege escalation.
### User Notes:
### Upgrade Notes:
- #### safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
  The safe_asterisk script now checks that, if it was run by the
  root user, the /etc/asterisk/startup.d directory and all the files it contains
  are owned by root.  If the checks fail, safe_asterisk will exit with an error
  and Asterisk will not be started.  Additionally, the default logging
  destination is now stderr instead of tty "9" which probably won't exist
  in modern systems.
### Developer Notes:
### Commit Authors:
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
## Issue and Commit Detail:
### Closed Issues:
  - !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in \ 
asterisk/res/res_stir_shaken/verification.c
  - !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk \ 
script may allow local privilege escalation.
### Commits By Author:
- #### George Joseph (1):
  - res_stir_shaken: Test for missing semicolon in Identity header.
- #### ThatTotallyRealMyth (1):
  - safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
### Commit List:
-  safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-  res_stir_shaken: Test for missing semicolon in Identity header.
### Commit Details:
#### safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
  Author: ThatTotallyRealMyth
  Date:   2025-06-10
  UpgradeNote: The safe_asterisk script now checks that, if it was run by the
  root user, the /etc/asterisk/startup.d directory and all the files it contains
  are owned by root.  If the checks fail, safe_asterisk will exit with an error
  and Asterisk will not be started.  Additionally, the default logging
  destination is now stderr instead of tty "9" which probably won't exist
  in modern systems.
  Resolves: #GHSA-v9q8-9j8m-5xwp
#### res_stir_shaken: Test for missing semicolon in Identity header.
  Author: George Joseph
  Date:   2025-07-31
  ast_stir_shaken_vs_verify() now makes sure there's a semicolon in
  the Identity header to prevent a possible segfault.
  Resolves: #GHSA-mrq5-74j5-f5cr

Log message:
Update to Asterisk 21.10.0:
## Change Log for Release asterisk-21.10.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
### Summary:
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
  - \ 
[GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2): \ 
cli_permissions.conf: deny option does not work for disallowing shell commands
### User Notes:
- #### res_stir_shaken.so: Handle X5U certificate chains.
  The STIR/SHAKEN verification process will now load a full
  certificate chain retrieved via the X5U URL instead of loading only
  the end user cert.
- #### res_stir_shaken: Add "ignore_sip_date_header" config option.
  A new STIR/SHAKEN verification option "ignore_sip_date_header" has
  been added that when set to true, will cause the verification process to
  not consider a missing or invalid SIP "Date" header to be a failure. \ 
 This
  will make the IAT the sole "truth" for Date in the verification process.
  The option can be set in the "verification" and "profile" \ 
sections of
  stir_shaken.conf.
  Also fixed a bug in the port match logic.
  Resolves: #1251
  Resolves: #1271
- #### app_record: Add RECORDING_INFO function.
  The RECORDING_INFO function can now be used
  to retrieve the duration of a recording.
- #### app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to \ 
raise penal..
  This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises
  only for members whose current penalty is within the [min_penalty, \ 
max_penalty] range.
  Members with lower or higher penalties are unaffected.
  This behavior is backward-compatible with existing queue rule configurations.
- #### res_odbc: cache_size option to limit the cached connections.
  New cache_size option for res_odbc to on a per class basis limit the
  number of cached connections. Please reference the sample configuration
  for details.
- #### res_odbc: cache_type option for res_odbc.
  When using res_odbc it should be noted that back-end
  connections to the underlying database can now be configured to re-use
  the cached connections in a round-robin manner rather than repeatedly
  re-using the same connection.  This helps to keep connections alive, and
  to purge dead connections from the system, thus more dynamically
  adjusting to actual load.  The downside is that one could keep too many
  connections active for a longer time resulting in resource also begin
  consumed on the database side.
- #### ARI Outbound Websockets
  Asterisk can now establish websocket sessions _to_ your ARI applications
  as well as accepting websocket sessions _from_ them.
  Full details: https://s.asterisk.net/ari-outbound-ws
- #### res_websocket_client: Create common utilities for websocket clients.
  A new module "res_websocket_client" and config file
  "websocket_client.conf" have been added to support several upcoming new
  capabilities that need common websocket client configuration.
- #### asterisk.c: Add option to restrict shell access from remote consoles.
  A new asterisk.conf option 'disable_remote_console_shell' has
  been added that, when set, will prevent remote consoles from executing
  shell commands using the '!' prefix.
  Resolves: #GHSA-c7p6-7mvq-8jq2
- #### sig_analog: Add Call Waiting Deluxe support.
  Call Waiting Deluxe can now be enabled for FXS channels
  by enabling its corresponding option.
### Upgrade Notes:
- #### jansson: Upgrade version to jansson 2.14.1
  jansson has been upgraded to 2.14.1. For more
  information visit jansson Github page: \ 
https://github.com/akheron/jansson/releases/tag/v2.14.1
  Resolves: #1178
- #### Alternate Channel Storage Backends
  With this release, you can now select an alternate channel
  storage backend based on C++ Maps.  Using the new backend may increase
  performance and reduce the chances of deadlocks on heavily loaded systems.
  For more information, see https://s.asterisk.net/dc679ec3
## Issue and Commit Detail:
### Closed Issues:
  - !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for \ 
disallowing shell commands
  - 271: [new-feature]: sig_analog: Add Call Waiting Deluxe support.
  - 548: [improvement]: Get Record() audio duration/length
  - 1088: [bug]: app_sms: Compilation failure in DEVMODE due to \ 
stringop-overflow error in GCC 15 pre-release
  - 1141: [bug]: res_pjsip: Contact header set incorrectly for call redirect \ 
(302 Moved temp.) when external_* set
  - 1178: [improvement]: jansson: Upgrade version to jansson 2.14.1
  - 1230: [bug]: ast_frame_adjust_volume and ast_frame_adjust_volume_float crash \ 
on interpolated frames
  - 1234: [bug]: Set CalllerID lost on DTMF attended transfer
  - 1240: [bug]: WebRTC invites failing on Chrome 136
  - 1243: [bug]: make menuconfig fails due to changes in GTK callbacks
  - 1251: [improvement]: PJSIP shouldn't require SIP Date header to process full \ 
shaken passport which includes iat
  - 1254: [bug]: ActiveChannels not reported when using AMI command PJSIPShowEndpoint
  - 1271: [bug]: STIR/SHAKEN not accepting port 8443 in certificate URLs
  - 1272: [improvement]: STIR/SHAKEN handle X5U certificate chains
  - 1276: MixMonitor produces broken recordings in bridged calls with asymmetric \ 
codecs (e.g., alaw vs G.722)
  - 1279: [bug]: regression: 20.12.0 downgrades quality of wav16 recordings
  - 1282: [bug]: Alternate Channel Storage Backends menuselect not enabling it
  - 1287: [bug]: channelstorage.c: Compilation failure with DEBUG_FD_LEAKS
  - 1288: [bug]: Crash when destroying channel with C++ alternative storage \ 
backend enabled
  - ASTERISK-30373: sig_analog: Add Call Waiting Deluxe options
### Commit List:
-  channelstorage: Rename callbacks that conflict with DEBUG_FD_LEAKS.
-  channelstorage_cpp_map_name_id: Fix callback returning non-matching channels.
-  channelstorage_makeopts.xml: Remove errant XML character.
-  res_stir_shaken.so: Handle X5U certificate chains.
-  res_stir_shaken: Add "ignore_sip_date_header" config option.
-  app_record: Add RECORDING_INFO function.
-  app_sms.c: Fix sending and receiving SMS messages in protocol 2
-  res_websocket_client:  Add more info to the XML documentation.
-  res_odbc: cache_size option to limit the cached connections.
-  res_odbc: cache_type option for res_odbc.
-  res_pjsip: Fix empty `ActiveChannels` property in AMI responses.
-  ARI Outbound Websockets
-  res_websocket_client: Create common utilities for websocket clients.
-  asterisk.c: Add option to restrict shell access from remote consoles.
-  frame.c: validate frame data length is less than samples when adjusting volume
-  res_audiosocket.c: Add retry mechanism for reading data from AudioSocket
-  res_audiosocket.c: Set the TCP_NODELAY socket option
-  menuselect: Fix GTK menu callbacks for Fedora 42 compatibility
-  jansson: Upgrade version to jansson 2.14.1
-  pjproject: Increase maximum SDP formats and attribute limits
-  manager.c: Invalid ref-counting when purging events
-  res_pjsip_nat.c: Do not overwrite transfer host
-  chan_pjsip: Serialize INVITE creation on DTMF attended transfer
-  sig_analog: Add Call Waiting Deluxe support.
-  app_sms: Ignore false positive vectorization warning.
-  lock.h: Add include for string.h when DEBUG_THREADS is defined.
-  Alternate Channel Storage Backends

Log message:
Upgrade to Asterisk 21.9.1:
pkgsrc changes:
- add gsed to USE_TOOLS
- appease pkglint somewhat
- PR/58978 -- comms/asterisk build fails if prefix is not /usr/pkg
## Change Log for Release asterisk-21.9.1
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
### Summary:
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
  - \ 
[GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw): \ 
Using malformed From header can forge identity with ";" or NULL in \ 
name portion
  - \ 
[GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2): \ 
cli_permissions.conf: deny option does not work for disallowing shell commands
### User Notes:
- #### asterisk.c: Add option to restrict shell access from remote consoles.
  A new asterisk.conf option 'disable_remote_console_shell' has
  been added that, when set, will prevent remote consoles from executing
  shell commands using the '!' prefix.
  Resolves: #GHSA-c7p6-7mvq-8jq2
### Commit Authors:
- George Joseph: (2)
## Issue and Commit Detail:
### Closed Issues:
  - !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with \ 
";" or NULL in name portion
  - !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for \ 
disallowing shell commands
### Commits By Author:
- #### George Joseph (2):
  - res_pjsip_messaging.c: Mask control characters in received From display name
  - asterisk.c: Add option to restrict shell access from remote consoles.

Log message:
Update to Asterisk 21.9.0.
## Change Log for Release asterisk-21.9.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
### Summary:
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
### User Notes:
- #### stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
  A Dial timeout on POST /channels/{channelId}/dial will now result in a
  CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer.  Previously
  no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.
- #### contrib: Add systemd service and timer files for malloc trim.
  Service and timer files for systemd have been added to the
  contrib/systemd/ directory. If you are experiencing memory issues,
  install these files to have "malloc trim" periodically run on the
  system.
- #### Add log-caller-id-name option to log Caller ID Name in queue log
  This patch adds a global configuration option, log-caller-id-name, to queues.conf
  to control whether the Caller ID name is logged as parameter 4 when a call \ 
enters a queue.
  When log-caller-id-name=yes, the Caller ID name is included in the queue log,
  Any '|' characters in the caller ID name will be replaced with '_'.
  (provided it’s allowed by the existing log_restricted_caller_id rules).
  When log-caller-id-name=no (the default), the Caller ID name is omitted.
- #### asterisk.c: Add "pre-init" and "pre-module" \ 
capability to cli.conf.
  In cli.conf, you can now define startup commands that run before
  core initialization and before module initialization.
- #### audiosocket: added support for DTMF frames
  The AudioSocket protocol now forwards DTMF frames with
  payload type 0x03. The payload is a 1-byte ascii representing the DTMF
  digit (0-9,*,#...).
### Upgrade Notes:
- #### ARI: REST over Websocket
  This commit adds the ability to make ARI REST requests over the same
  websocket used to receive events.
  See \ 
https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
### Commit Authors:
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)
## Issue and Commit Detail:
### Closed Issues:
  - 505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs \ 
created by ast_sockaddr_from_pj_sockaddr()
  - 643: [new-feature]: pjsip show contact -- show all details same as AMI \ 
PJSIPShowContacts
  - 963: [bug]: missing hangup cause for ARI ChannelDestroyed when Dial times out
  - 1091: [improvement]: app queue :add to  queue log callerid name
  - 1144: [bug]: action_redirect don't remove bridge_after_goto data
  - 1171: [improvement]: Need the capability in audiohook.c for fractional \ 
(float) type volume adjustments.
  - 1181: [bug]: Incorrect PJSIP Endpoint Device States on Multiple Channels
  - 1190: [bug]: Crash when starting ConfBridge recording over CLI and AMI
  - 1197: [bug]: ChannelHangupRequest does not show cause code in all cases
  - 1206: [improvement]: chan_iax2: Minor improvements to documentation and \ 
warning messages.
  - 1220: [bug]: res_pjsip_caller_id: OLI is not parsed if contained in a URI \ 
parameter
  - 1224: [improvement]: app_meetme: Removal version is incorrect
### Commit List:
-  res_pjsip_caller_id: Also parse URI parameters for ANI2.
-  app_meetme: Remove inaccurate removal version from xmldocs.
-  docs: Fix typos in apps/
-  stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-  chan_iax2: Minor improvements to documentation and warning messages.
-  pbx_ael: unregister AELSub application and CLI commands on module load failure
-  res_pjproject: Fix DTLS client check failing on some platforms
-  Prequisites for ARI Outbound Websockets
-  contrib: Add systemd service and timer files for malloc trim.
-  action_redirect: remove after_bridge_goto_info
-  channel: Always provide cause code in ChannelHangupRequest.
-  Add log-caller-id-name option to log Caller ID Name in queue log
-  asterisk.c: Add "pre-init" and "pre-module" capability to \ 
cli.conf.
-  app_confbridge: Prevent crash when publishing channel-less event.
-  ari_websockets: Fix frack if ARI config fails to load.
-  ARI: REST over Websocket
-  audiohook.c: Add ability to adjust volume with float
-  audiosocket: added support for DTMF frames
-  asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()'
-  audiosocket: fix timeout, fix dialplan app exit, server address in logs
-  Update config.guess and config.sub
-  chan_pjsip: set correct Endpoint Device State on multiple channels
-  file.c: missing "custom" sound files should not generate warning logs
## Change Log for Release asterisk-21.8.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
### Summary:
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
### User Notes:
- #### ari/pjsip: Make it possible to control transfers through ARI
  Call transfers on the PJSIP channel can now be controlled by
  ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
  dialplan function.
### Commit Authors:
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)
## Issue and Commit Detail:
### Closed Issues:
  - 211: [bug]: stasis: Off-nominal channel leave causes bridge to be destroyed
  - 1085: [bug]: utils: Compilation failure with DEVMODE due to old-style definitions
  - 1101: [bug]: when setting a  var with a double quotes and using Set(HASH)
  - 1109: [bug]: Off nominal memory leak in res/ari/resource_channels.c
  - 1112: [bug]: STIR/SHAKEN verification doesn't allow anonymous callerid to be \ 
passed to the dialplan.
  - 1119: [bug]: Realtime database not working after upgrade from 22.0.0 to 22.2.0
  - 1122: Need status on CVE-2024-57520 claim.
  - 1124: [bug]: Race condition between bridge and channel delete can over-write \ 
cause code set in hangup.
  - 1131: [bug]: CHANGES link broken in README.md
  - 1135: [bug]: Problems with video decoding due to RTP marker bit set
  - 1149: [bug]: res_pjsip: Mismatch in tcp_keepalive_enable causes not to enable
  - 1164: [bug]: WARNING Message in messages.log for res_curl.conf [globals]
### Commit List:
-  documentation: Update Gosub, Goto, and add new documentationtype.
-  res_config_curl.c: Remove unnecessary warnings.
-  README.md: Updates and Fixes
-  res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`.
-  fix: Correct default flag for tcp_keepalive_enable option
-  docs: AMI documentation fixes.
-  config.c: #include of non-existent file should not crash
-  manager.c: Check for restricted file in action_createconfig.
-  swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().
-  Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too \ 
big"
-  res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer.
-  docs: Fix typos in cdr/ Found via codespell
-  bridging: Fix multiple bridging issues causing SEGVs and FRACKs.
-  res_config_pgsql: Fix regression that removed dbname config.
-  res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.
-  resource_channels.c: Fix memory leak in ast_ari_channels_external_media.
-  ari/pjsip: Make it possible to control transfers through ARI
-  channel.c: Remove dead AST_GENERATOR_FD code.
-  func_strings.c: Prevent SEGV in HASH single-argument mode.
-  docs: Add version information to AGI command XML elements.
-  docs: Fix minor typo in MixMonitor AMI action
-  utils: Disable old style definition warnings for libdb.
-  rtp.conf.sample: Correct stunaddr example.
-  docs: Add version information to ARI resources and methods.
-  docs: Indent <since> tags.
## Change Log for Release asterisk-21.7.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
### Summary:
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
### User Notes:
- #### sig_analog: Add Last Number Redial feature.
  Users can now redial the last number
  called if the lastnumredial setting is set to yes.
  Resolves: #437
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
  The SHA-256 and SHA-512-256 algorithms are now available
  for authentication as both a UAS and a UAC.
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
  Bundled pjproject has been upgraded to 2.15.1. For more
  information visit pjproject Github page: \ 
https://github.com/pjsip/pjproject/releases/tag/2.15.1
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
  The pjsip.conf AOR section now has a "qualify_2xx_only"
  option that can be set so that only 2XX responses to OPTIONS requests
  used to qualify a contact will mark the contact as available.
- #### app_queue: allow dynamically adding a queue member in paused state.
  use the p option of AddQueueMember() for paused member state.
  Optionally, use the r(reason) option to specify a custom reason for the pause.
- #### manager.c: Add Processed Call Count to CoreStatus output
  The current processed call count is now returned as CoreProcessedCalls from the
  CoreStatus AMI Action.
- #### func_curl.c: Add additional CURL options for SSL requests
  The following new configuration options are now available
  in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost'
  (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath'
  (CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype'
  (CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
  (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the
  libcurl documentation for more details.
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
  You can now set the "unknown_tn_attest_level" option
  in the attestation and/or profile objects in stir_shaken.conf to
  enable sending Identity headers for callerid TNs not explicitly
  configured.
### Upgrade Notes:
- #### alembic: Database updates required.
  Two commits in this release...
  'Add SHA-256 and SHA-512-256 as authentication digest algorithms'
  'res_pjsip: Add new AOR option "qualify_2xx_only"'
  ...have modified alembic scripts for the following database tables: ps_aors,
  ps_contacts, ps_auths, ps_globals. If you don't use the scripts to update
  your database, reads from those tables will succeeed but inserts into the
  ps_contacts table by res_pjsip_registrar will fail.
### Commit Authors:
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)
## Issue and Commit Detail:
### Closed Issues:
  - 437: [new-feature]: sig_analog: Add Last Number Redial
  - 851: [bug]: unable to read audiohook both side when packet lost on one side \ 
of the call
  - 921: [bug]: Stir-Shaken doesn’t allow B or C attestation for unknown \ 
callerid which is allowed by ATIS-1000074.v003, §5.2.4
  - 927: [bug]: no audio when media source changed during the call
  - 948: [improvement]: Support SHA-256 algorithm on REGISTER and INVITE challenges
  - 993: [bug]: sig_analog: Feature Group D / E911 no longer work
  - 999: [bug]: Crash when setting a global variable with invalid UTF8 characters
  - 1007: [improvement]: Cannot dynamically add queue member in paused state \ 
from dialplan or command line
  - 1013: [improvement]: chan_pjsip: Send VIDUPDATE RTP frames for H.264 streams \ 
on endpoints without WebRTC
  - 1021: [improvement]: proper queue_log paused state when member added dynamically
  - 1023: [improvement]: Improve PJSIP_MEDIA_OFFER documentation
  - 1028: [bug]: "pjsip show endpoints" shows some identifies on \ 
endpoints that shouldn't be there
  - 1029: [bug]: chan_dahdi: Wrong channel state set when RINGING received
  - 1054: [bug]: chan_iax2: Frames unnecessarily backlogged with jitterbuffer if \ 
no voice frames have been received yet
  - 1058: [bug]: Asterisk fails to compile following commit 71a2e8c on Ubuntu 20.04
  - 1064: [improvement]: ast_tls_script: Add option to skip passphrase for CA \ 
private key
  - 1075: [bug]: res_prometheus does not set Content-Type header in HTTP response
  - 1095: [bug]: res_pjsip missing "Failed to authenticate" log entry \ 
for unknown endpoint
  - 1097: [bug]: res_pjsip/pjsip_options. ODBC: Unknown column 'qualify_2xx_only'
### Commit List:
-  res_pjsip_authenticator_digest: Make correct error messages appear again.
-  alembic: Database updates required.
-  res_pjsip: Fix startup/reload memory leak in config_auth.
-  docs: Add version information to application and function XML elements
-  docs: Add version information to manager event instance XML elements
-  LICENSE: Update company name, email, and address.
-  res_prometheus.c: Set Content-Type header on /metrics response.
-  README.md, asterisk.c: Update Copyright Dates
-  docs: Add version information to configObject and configOption XML elements
-  res_pjsip_authenticator_digest: Fix issue with missing auth and DONT_OPTIMIZE
-  ast_tls_cert: Add option to skip passphrase for CA private key.
-  chan_iax2: Avoid unnecessarily backlogging non-voice frames.
-  config.c: fix #tryinclude being converted to #include on rewrite
-  sig_analog: Add Last Number Redial feature.
-  docs: Various XML fixes
-  strings.c: Improve numeric detection in `ast_strings_match()`.
-  docs: Enable since/version handling for XML, CLI and ARI documentation
-  logger.h: Fix build when AST_DEVMODE is not defined.
-  dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's `media` argument.
-  samples: Use "asterisk" instead of "postgres" for username
-  manager: Add `<since>` tags for all AMI actions.
-  logger.c fix: malformed JSON template
-  manager.c: Rename restrictedFile to is_restricted_file.
-  res_pjproject: Fix typo (OpenmSSL->OpenSSL)
-  Add SHA-256 and SHA-512-256 as authentication digest algorithms
-  config.c: retain leading whitespace before comments
-  config.c: Fix off-nominal reference leak.
-  normalize contrib/ast-db-manage/queue_log.ini.sample
-  Add C++ Standard detection to configure and fix a new C++20 compile issue
-  chan_dahdi: Fix wrong channel state when RINGING recieved.
-  Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-  gcc14: Fix issues caught by gcc 14
-  Header fixes for compiling C++ source files
-  Add ability to pass arguments to unit tests from the CLI
-  res_pjsip: Add new AOR option "qualify_2xx_only"
-  res_odbc: release threads from potential starvation.
-  Allow C++ source files (as extension .cc) in the main directory
-  format_gsm.c: Added mime type
-  func_uuid: Add a new dialplan function to generate UUIDs
-  app_queue: allow dynamically adding a queue member in paused state.
-  chan_iax2: Add log message for rejected calls.
-  chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams
-  res_curl.conf.sample: clean up sample configuration and add new SSL options
-  res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big
-  res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF
-  manager.c: Add Processed Call Count to CoreStatus output
-  func_curl.c: Add additional CURL options for SSL requests
-  sig_analog: Fix regression with FGD and E911 signaling.
-  res_stir_shaken: Allow sending Identity headers for unknown TNs
## Change Log for Release asterisk-21.6.1
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
### Summary:
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - \ 
[GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh): \ 
Path traversal via AMI ListCategories allows access to outside files
### User Notes:
- #### manager.c: Restrict ListCategories to the configuration directory.
  The ListCategories AMI action now restricts files to the
  configured configuration directory.
### Commit Authors:
- Ben Ford: (1)
## Issue and Commit Detail:
### Closed Issues:
  - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to \ 
outside files
### Commit List:
-  manager.c: Restrict ListCategories to the configuration directory.
### Commit Details:
#### manager.c: Restrict ListCategories to the configuration directory.
  Author: Ben Ford
  Date:   2024-12-17
  When using the ListCategories AMI action, it was possible to traverse
  upwards through the directories to files outside of the configured
  configuration directory. This action is now restricted to the configured
  directory and an error will now be returned if the specified file is
  outside of this limitation.
  Resolves: #GHSA-33x6-fj46-6rfh
  UserNote: The ListCategories AMI action now restricts files to the
  configured configuration directory.
## Change Log for Release asterisk-21.6.0
### Links:
 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
### Summary:
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
### User Notes:
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
  The new "suppress_moh_on_sendonly" endpoint option
  can be used to prevent playing MOH back to a caller if the remote
  end sends "sendonly" or "inactive" (hold) to Asterisk in \ 
an SDP.
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
  The MixMonitor application now has a new 'D' option which
  interleaves the recorded audio in the output frames. This allows for
  stereo recording output with one channel being the transmitted audio and
  the other being the received audio. The 't' and 't' options are
  compatible with this.
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
  The ModuleLoad AMI action now restricts modules to the
  configured modules directory.
- #### manager: Enhance event filtering for performance
  You can now perform more granular filtering on events
  in manager.conf using expressions like
  `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
  This is much more efficient than
  `eventfilter = Event: Newchannel.*Channel: PJSIP/`
  Full syntax guide is in configs/samples/manager.conf.sample.
- #### db.c: Remove limit on family/key length
  The `ast_db_*()` APIs have had the 253 byte limit on
  "/family/key" removed and will now accept families and keys with a
  total length of up to SQLITE_MAX_LENGTH (currently 1e9!).  This
  affects the `DB*` dialplan applications, dialplan functions,
  manager actions and `databse` CLI commands.  Since the
  media_cache also uses the `ast_db_*()` APIs, you can now store
  resources with URIs longer than 253 bytes.
### Upgrade Notes:
### Commit Authors:
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)
## Issue and Commit Detail:
### Closed Issues:
  - 487: [bug]: Segfault possibly in ast_rtp_stop
  - 821: [bug]: app_dial:  The progress timeout doesn't cause Dial to exit
  - 881: [bug]: Long URLs are being rejected by the media cache because of an \ 
astdb key length limit
  - 882: [bug]: Value CHANNEL(userfield) is lost by BRIDGE_ENTER
  - 897: [improvement]: Restrict ModuleLoad AMI action to the modules directory
  - 900: [bug]: astfd.c: NULL pointer passed to fclose with nonnull attribute \ 
causes compilation failure
  - 902: [bug]: app_voicemail: Pager emails are ill-formatted when custom \ 
subject is used
  - 916: [bug]: Compilation errors on FreeBSD
  - 923: [bug]: Transport monitor shutdown callback only works on the first \ 
disconnection
  - 924: [bug]: dnsmgr.c: dnsmgr_refresh() should not flag change if IP address \ 
order changes
  - 928: [bug]: chan_dahdi: MWI while off-hook when hung up on after recall ring
  - 932: [bug]: When connected to multiple IP addresses the transport monitor is \ 
only set on the first one
  - 937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'
  - 938: [bug]: memory leak - CBAnn leaks a small amount format_cap related \ 
memory for every confbridge
  - 945: [improvement]: Add stereo recording support for app_mixmonitor
  - 951: [new-feature]: func_evalexten: Add `EVAL_SUB` function
  - 974: [improvement]: change and/or remove some wiki mentions to docs mentions \ 
in the sample configs
  - 979: [improvement]: Add ability to suppress MOH when a remote endpoint sends \ 
"sendonly" or "inactive"
  - 982: [bug]: The addition of tenantid to the ast_sip_endpoint structure broke \ 
ABI compatibility
  - 990: [improvement]: The help for PJSIP_AOR should indicate that you need to \ 
call PJSIP_CONTACT to get contact details
  - 995: [bug]: suppress_moh_on_sendonly should use AST_BOOL_VALUES instead of \ 
YESNO_VALUES in alembic script
### Commit List:
-  res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T
-  res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-  res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
-  samples: remove and/or change some wiki mentions
-  func_pjsip_aor/contact: Fix documentation for contact ID
-  res_pjsip: Move tenantid to end of ast_sip_endpoint
-  pjsip_transport_events: handle multiple addresses for a domain
-  func_evalexten: Add EVAL_SUB function.
-  res_srtp: Change Unsupported crypto suite msg from verbose to debug
-  Add res_pjsip_config_sangoma external module.
-  app_mixmonitor: Add 'D' option for dual-channel audio.
-  pjsip_transport_events: Avoid monitor destruction
-  app_dial: Fix progress timeout calculation with no answer timeout.
-  pjproject_bundled:  Tweaks to support out-of-tree development
-  core_unreal.c: Fix memory leak in ast_unreal_new_channels()
-  dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
-  geolocation.sample.conf: Fix comment marker at end of file
-  func_base64.c: Ensure we set aside enough room for base64 encoded data.
-  app_dial: Fix progress timeout.
-  chan_dahdi: Never send MWI while off-hook.
-  manager.c: Add unit test for Originate app and appdata permissions
-  alembic: Drop redundant voicemail_messages index.
-  res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
-  main, res, tests: Fix compilation errors on FreeBSD.
-  res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
-  manager.c: Restrict ModuleLoad to the configured modules directory.
-  res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
-  cdr_custom: Allow absolute filenames.
-  astfd.c: Avoid calling fclose with NULL argument.
-  channel: Preserve CHANNEL(userfield) on masquerade.
-  cel_custom: Allow absolute filenames.
-  app_voicemail: Fix ill-formatted pager emails with custom subject.
-  res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
-  Fix application references to Background
-  manager.conf.sample: Fix mathcing typo
-  manager: Enhance event filtering for performance
-  manager.c: Split XML documentation to manager_doc.xml
-  db.c: Remove limit on family/key length