Log message:
*: Recursive revbump from security/nettle-4.0

Log message:
*: remove OWNER definition

OWNER, when it was introduced, was to protect packages deep in the
infrastructure by emphasizing that they should not be touched by
non-MAINTAINERs.

No infrastructure package still sets OWNER.

Note: non-trivial change to packages should be passed by MAINTAINERs.

As discussed on tech-pkg.

Log message:
Update to Asterisk 21.12.2:

Security update for PJSIP vulnerabilities.

## Change Log for Release asterisk-21.12.2

### Links:

 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0

## Issue and Commit Detail:

### Closed Issues:

 - 1833: [bug]: Address security vulnerabilities in pjproject

### Commit List:

- res_pjsip: Address pjproject security vulnerabilities

### Commit Details:

#### res_pjsip: Address pjproject security vulnerabilities
 Author: Mike Bradeen
 Date: 2026-03-25

 Address the following pjproject security vulnerabilities

 [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long \ 
username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
 [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription \ 
termination \ 
header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
 [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race \ 
conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
 [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart \ 
parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)

 Resolves: #1833

Log message:
update to Asterisk 21.12.1: this is a security fix

## Change Log for Release asterisk-21.12.1

### Links:

 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)

### Summary:

- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
 - \ 
[GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42): \ 
Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
 - \ 
[GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3): \ 
ast_coredumper running as root sources ast_debug_tools.conf from /etc/asterisk; \ 
potentially leading to privilege escalation
 - \ 
[GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh): \ 
The Asterisk embedded web server's /httpstatus page echos user supplied \ 
values(cookie and query string) without sanitization
 - \ 
[GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c): \ 
ast_coredumper runs as root, and writes gdb init file to world writeable folder; \ 
leading to potential privilege escalation

### User Notes:

- #### ast_coredumper: check ast_debug_tools.conf permissions
 ast_debug_tools.conf must be owned by root and not be
 writable by other users or groups to be used by ast_coredumper or
 by ast_logescalator or ast_loggrabber when run as root.

### Upgrade Notes:

- #### http.c: Change httpstatus to default disabled and sanitize output.
 To prevent possible security issues, the `/httpstatus` page
 served by the internal web server is now disabled by default. To explicitly
 enable it, set `enable_status=yes` in http.conf.

## Issue and Commit Detail:

### Closed Issues:

 - !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to \ 
potential XXE Injection
 - !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources \ 
ast_debug_tools.conf from /etc/asterisk; potentially leading to privilege \ 
escalation
 - !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page \ 
echos user supplied values(cookie and query string) without sanitization
 - !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file \ 
to world writeable folder; leading to potential privilege escalation

### Commits By Author:

- #### George Joseph (2):

- #### Mike Bradeen (2):

### Commit List:

- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions

Log message:
*: recursive bump for nettle 4.0 shlib major bump

Log message:
*: recursive bump for icu 78.1

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 http://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