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comment	@# @;


1.5
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1.5
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@Update to Asterisk 23.4.0:


## Change Log for Release asterisk-23.4.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)

### Summary:

- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0

### User Notes:

- #### res ari: Add attachable states to Channels and Bridges
  Bridge variables now can be set and retrieved via the following paths:
  `/bridges/{bridgeId}/variable`
  `/bridges/{bridgeId}/variables`
  Both Bridge and Channel variables can now be set with an optional 'report_events'
  boolean flag that will cause those variables to be included on all events on that
  object. The 'report_events' flag will default to False if not set to maintain
  backwards capability.
  To allow this, variables can now be either name value pairs (the current format):
  `<variable_name>: '<value_string>'`
   - or -
  `<variable_name>: {value: '<value_string>', report_events: [true|false]}`

- #### ARI: Added paths to get and set multiple channel variables.
  Added new ARI paths for getting and setting multiple channel
  variables at a time. For GET, this takes in a single string of
  comma-separated variable names, while POST takes in a dictionary of key
  value pairs. The behavior is the same as passing in variables when
  originating a channel.

- #### res_rtp_asterisk: Add option to control stun host resolution when TTL = 0
  A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf
  that allows control over what happens when a STUN server hostname lookup
  returns a TTL of 0.  The values can be set as follows:
  - 'no': This is the historical (and current default) behavior of not doing
  any further lookups and continuing to use the last successful result until
  Asterisk is restarted or rtp.conf is reloaded.
  - 'yes': Use the last cached result for the current call but trigger
  re-resolution in the background for the benefit of future calls.
  If the result of the background lookup is a ttl > 0, periodic resolution
  will be restarted otherwise the next call will use the new cached value
  and will trigger a background lookup again.
  A new CLI command `rtp resolve stun hostname` has been added
- #### app_dial: Properly handle callee hangup while sending digits.
  If a called channel sends progress or wink and the caller begins
  sending digits but the callee answers and then hangs up before digit
  sending can finish, the call is now answered before being disconnected.
  If the callee hangs up without answering, the call now continues in
  the dialplan.

- #### Upgrade bundled pjproject to 2.17.
  Bundled pjproject has been upgraded to 2.17. For more
  information about what is included in this release, see the
  pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17

- #### res_pjsip: Add per-endpoint RTP port range configuration
  PJSIP endpoints now support rtp_port_start and
  rtp_port_end options to configure a dedicated RTP port range per
  endpoint, overriding the global rtp.conf setting.

- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins
  New optional modules res_stasis_broadcast.so and
  app_stasis_broadcast.so enable broadcasting an incoming channel to multiple
  ARI applications. The first application to successfully claim (via
  POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan
  application initiates broadcasts. CallBroadcast and CallClaimed events notify
  applications. When modules are not loaded, behavior is unchanged.

- #### chan_iax2: Add CHANNEL getter to retrieve auth method.
  CHANNEL(auth_method) can now be used to retrieve the
  auth method negotiated for a call on IAX2 channels.

- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
  New module res_pjsip_maintenance adds runtime maintenance
  mode for PJSIP endpoints. Use "pjsip set maintenance <on|off>
  <endpoint|all>" to enable or disable, and "pjsip show maintenance"
  to list affected endpoints. AMI actions PJSIPSetMaintenance and
  PJSIPShowMaintenance provide programmatic access. No configuration
  file changes required.


### Upgrade Notes:

- #### jansson: Upgrade version to jansson 2.15.0
  jansson has been upgraded to 2.15.0. For more
  information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0

- #### res_pjsip: Add per-endpoint RTP port range configuration
  An alembic database migration has been added to add
  the rtp_port_start and rtp_port_end columns to the ps_endpoints
  table. Run "alembic upgrade head" to apply the schema change.


### Developer Notes:

- #### res_pjsip: Add per-endpoint RTP port range configuration
  New public API: ast_rtp_instance_new_with_port_range()
  creates an RTP instance with a per-instance port range.
  ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end()
  allow RTP engines to query the override. Third-party RTP engines can
  use these getters to support per-instance port ranges.

- #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins
  New public APIs in stasis_app_broadcast.h:
  stasis_app_broadcast_channel(), stasis_app_claim_channel(),
  stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event
  types (CallBroadcast, CallClaimed) added to events.json. All code is isolated;
  no existing ABI modified.

- #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
  ast_sip_session_supplement gains a new optional
  callback - int (*session_create)(struct ast_sip_endpoint *endpoint,
  const char *destination). It is called from the global supplement
  list (not per-session) at the start of ast_sip_session_create_outgoing()
  via ast_sip_session_check_supplement_create(). Returning non-zero
  blocks the outgoing session. Modules that need to gate outbound
  SIP session creation should register a supplement with this callback
  set rather than hooking into chan_pjsip directly.

- #### build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build
  The pjsua and pjsystest application binaries, the deprecated
  Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub
  implementations are no longer built or installed as part of the bundled
  pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency
  is also removed. Developers who relied on the pjsua binary for Test Suite
  SIP simulation should use SIPp instead, which is the current Asterisk Test
  Suite standard.
  Fixes: #1840


## Issue and Commit Detail:

### Closed Issues:

  - 1217: [bug]: INSERT INTO cdr query prepare statement issue on cdr_adaptive_odbc to control statement preparation manually
  - 1357: [bug]: MessageSend WARNING “not a valid SIP/SIPS URI” when using endpoint not URI
  - 1653: [bug]: Asterisk ODBC Voicemail Crash Caused by Voicemail Re-entry Loop and Unsafe BLOB Retrieval
  - 1736: app_queue: update_queue() may double-increment member->calls with shared_lastcall=yes (regression observed after 20.17; impacts fewestcalls routing)
  - 1761: func_talkdetect.c: TALK_DETECT docs wording mistake
  - 1762: [bug]: 100% CPU usage when entering BridgeWait after JITTERBUFFER(disabled)=
  - 1807: [new-feature]: translate.c: implement different types of sample frame inputs
  - 1812: [new-feature]: add tests/test_codec_translations.c
  - 1818: [bug]: func_odbc: possible use-after-free crash during reload with active calls
  - 1839: Crash in MDMF Caller ID parser due to signed char length field on DAHDI channels
  - 1840: [bug]: Asterisk fails to compile with --enable-dev-mode=yes due to INIT_RETURN undeclared in bundled pjproject Python bindings
  - 1855: [bug]: core reload deadlocks Asterisk (pjsip, CLI, etc.)
  - 1858: [bug]: DNS records with a TTL of zero are permanently cached
  - 1859: [bug]: res_pjsip_outbound_registration: No expires header set when triggered via CLI
  - 1861: [bug]: Possible heap corruption in audiohook/translate write path during bridged media
  - 1862: [bug]: Build fails with Building Documentation: line 210: /tmp/xmldoc.tmp.xml: Permission denied
  - 1865: [bug]: chan_iax2: Another code path that causes crashes on negative data lengths
  - 1867: [bug]: Massive [eventpoll] file-descriptor leak (hundreds of epoll fds) when TURN is enabled in rtp.conf
  - 1872: [bug]:  Deadlock in chan_pjsip_new when endpoint set_var invokes PJSIP_HEADER
  - 1878: [new-feature]: chan_iax2: Allow retrieving the auth method using the CHANNEL function
  - 1883: [bug]: fix: stdatomic.h false positive on GCC 4.8
  - 1885: [bug]: cdrel_custom :SQLite version too old: sqlite3_prepare_v3 / SQLITE_PREPARE_PERSISTENT undeclared
  - 1888: [improvement]: pjsip: Upgrade bundled version to pjproject 2.17
  - 1892: [bug]: Build failure with bundled pjproject on OpenSSL 1.0.x: undefined reference to TLS_method and SSL_CTX_set_ciphersuites
  - 1894: [bug]: Outbound ARI websockets don't always clean up completely
  - 1896: [bug]: asterisk.c fails to compile when HAVE_LIBEDIT_IS_UNICODE isn't defined
  - 1901: [bug]: QUEUE_RAISE_PENALTY=rN ignored when set via queue rules
  - 1903: [bug]: g++ 16 no longer defines __STDC_VERSION__ causing channelstorage_cpp_map_name_id.cc to fail
  - 1907: [bug]: Deadlock between bridge and setting of RTP stats variables at hangup
  - 1910: [improvement]: Add attachable state variables to Channels and Bridges.
  - 1915: [bug]: app_dial: Channel not handled properly if callee disconnects while caller is sending it digits prior to answer
  - 1921: [bug]: Memory error in crypto_get_cert_subject when using malloc_debug
  - 1928: [bug]: Calling ast_softhangup with channel lock held can cause deadlock
  - 1931: [improvement]: jansson: Upgrade version to jansson 2.15.0
  - 1936: [bug]: Calling set_variable on PJSIP channel when originating with ARI with PJSIP_HEADER can result in deadlock
  - 1938: [bug]: res_rtp_asterisk: Copy/paste error in ast_rtp_get_stat()
  - 1941: [bug]: chan_websocket doesn't handle CONTINUATION websocket frames
  - 1947: [bug]: chan_dahdi fails to build with gcc-16 when openr2 is installed
  - 1950: [bug]: app_record does not detect channel hangup during beep playback
  - 1952: [bug]: OpenSSL 4.0.0
  - 1957: [bug]: Calendar module fails to build with libical 4.X
  - 1970: [bug]: Startup or shutdown segfault in res_ari_model under certain conditions with DEVMODE and persistent outbound websockets.

### Commit List:

-  res_ari: Add res_ari_model as an optional_module.
-  res ari: Add attachable states to Channels and Bridges
-  ARI: Added paths to get and set multiple channel variables.
-  res_stir_shaken: avoid direct ASN1_STRING accesses
-  tcptls.c: fix build with OpenSSL 4
-  res_calendar: Fix build with libical 4.X
-  app_record: Fix hangup handling during beep playback
-  odbc: Don't use prepared statements for distinct SQL statements
-  abstract_jb.c: Remove timerfd from channel when disabling jitter buffer
-  res_pjsip: Don't allow a leading period when wildcard matching
-  Ensure channel locks aren't held while calling ast_set_variables.
-  app_queue: fix double increment of member->calls with shared_lastcall
-  chan_dahdi: Fix set but not used in mfcr2_show_links_of().
-  tests: add tests/test_codec_translations.c
-  install_prereq: Add a 'minimal' mode for basic build dependencies
-  chan_websocket: Handle incoming CONTINUATION frames.
-  res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat().
-  jansson: Upgrade version to jansson 2.15.0
-  channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup.
-  res_rtp_asterisk: Add option to control stun host resolution when TTL = 0
-  pjsip_configuration: Show actual dtls_verify config.
-  app_dial: Properly handle callee hangup while sending digits.
-  res_pjsip_messaging: Update To URI only if it is a SIP(S) URI
-  Upgrade bundled pjproject to 2.17.
-  res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug
-  manager: Eliminate unnecessary code, simplify sessions in stasis callbacks
-  res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup
-  res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage
-  channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars
-  chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER
-  res_pjsip: Add per-endpoint RTP port range configuration
-  app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules
-  app_voicemail_odbc: fix msgnum race and crash on failed STORE
-  ari_websockets: Fix two issues in the cleanup of outbound websockets.
-  compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++.
-  pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c
-  asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE.
-  cdrel_custom: fix SQLite compatibility for versions < 3.20.0
-  translate.c: implement different sample_types for translation computation.
-  stasis_broadcast: Add optional ARI broadcast with first-claim-wins
-  res_audiosocket: Tolerate non-audio frame types
-  pbx_functions: Save module pointer before calling read and write callbacks.
-  chan_iax2: Add CHANNEL getter to retrieve auth method.
-  fix: backport pjproject stdatomic.h GCC 4.8 build failure patch
-  res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy.
-  res_pjsip_maintenance: Add PJSIP endpoint maintenance mode
-  chan_iax2: Add another check to abort frame handling if datalen < 0.
-  res_pjsip_outbound_registration: only update the Expires header if the value has changed
-  func_talkdetect.c: Clarify dsp_talking_threshold documentation.
-  make_xml_documentation: Remove temporary file on script exit.
-  res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread
-  build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build
-  callerid: fix signed char causing crash in MDMF parser
@
text
@$NetBSD: distinfo,v 1.4 2026/04/13 03:47:11 jnemeth Exp $

BLAKE2s (asterisk-23.4.0/asterisk-23.4.0.tar.gz) = 4dacf1403121ee6e897c851c815b780d5ad91685ad68e095655a09f7b4696ce2
SHA512 (asterisk-23.4.0/asterisk-23.4.0.tar.gz) = da50d6d26e631e6a5e90d6eb5efa78c46fb9e852eb0bcfe01b7e82332e3361b024bd9a1e63d9c5f21d56cd3e3817f28b1a97d7be7f2194e0d506560d3a07071c
Size (asterisk-23.4.0/asterisk-23.4.0.tar.gz) = 26525287 bytes
BLAKE2s (asterisk-23.4.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-23.4.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-23.4.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-23.4.0/pjproject-2.17.md5) = 40b4cd26a5101f81b6d57f5d8b8a74fe4b296292b21677e02a1bcc73d0d3a3c3
SHA512 (asterisk-23.4.0/pjproject-2.17.md5) = 8946eebc9d7d472e001d03602a8a6b12632734dbc665acb5aca60c0f83ebbd4d69d31c06096fde44907ce2040be5fd4145ff7fcaaa2f681f90600faa0ed5f294
Size (asterisk-23.4.0/pjproject-2.17.md5) = 166 bytes
BLAKE2s (asterisk-23.4.0/pjproject-2.17.tar.bz2) = 3909b745c89a119f41126ee49ea196d5bf6889fdcbeb22d266dabc3fd339dd96
SHA512 (asterisk-23.4.0/pjproject-2.17.tar.bz2) = b523b3761aab8931ad2a940d107861caf346e3c5cf9b12d23b5b267b8f3b4758ee842b401f1199c3ceb7e78bbaf0d2a167d2574222c51e6016586eb2caa1e33d
Size (asterisk-23.4.0/pjproject-2.17.tar.bz2) = 8975382 bytes
SHA1 (patch-Makefile) = 5cf3b6937ec23a82e4d056b91e493a36bc1089b9
SHA1 (patch-addons_chan__ooh323.c) = 1775da7ca2129a962ed460bd1e78ba3ce6afa62c
SHA1 (patch-apps_app__adsiprog.c) = 031139e5cd1ef6bb2afb0a74fee3d752eded0a2c
SHA1 (patch-apps_app__chanspy.c) = 29a807909645c1ad0c8f81b6513a284b978e7c47
SHA1 (patch-apps_app__directory.c) = 889a78123033709d28b0b805f2a379242ccd7dcc
SHA1 (patch-apps_app__dumpchan.c) = 127ac02bdc180ad2334cd095aa6e646feb6fba10
SHA1 (patch-apps_app__followme.c) = c6a5790b5e9b34d07dbfdd66a58e2854c8c72695
SHA1 (patch-apps_app__minivm.c) = 22ee6ebfbe205baf0acf46ab16c94fea1750f2fb
SHA1 (patch-apps_app__queue.c) = fdf7cf202b60e24cd9227f7e461bbd541565d602
SHA1 (patch-apps_app__sms.c) = ad65b3cb2a30489551101f7534c691cd1155d18f
SHA1 (patch-apps_app__voicemail.c) = 5276457466fde27494bf43fd6d306397bc4ff97f
SHA1 (patch-build__tools_make__xml__documentation) = c0552e7376fe7c4c12b9fb2cb20ca1a67997872d
SHA1 (patch-build__tools_mkpkgconfig) = 7fab8fcf46d9f8a3b98455674fec6307ec472b23
SHA1 (patch-cdr_cdr__pgsql.c) = 82b002a1f5ed3b7361a98e2bffb5cea8833949b8
SHA1 (patch-cel_cel__pgsql.c) = b280efab2b035ce60be268bac9bc8824910b2b8f
SHA1 (patch-channels_chan__pjsip.c) = efd4cbb82133fc5ddf7de70d01c99e185c585211
SHA1 (patch-channels_pjsip_cli__commands.c) = 01baa9d242e3af02a1f3540cfb3064ad68c71d67
SHA1 (patch-channels_pjsip_dialplan__functions.c) = 2cf8199c4ec9d4894eb922c2703d49ecc06188ef
SHA1 (patch-configure) = 3a489e0c508cb2a46e8ada6801369756c16d1dea
SHA1 (patch-configure.ac) = 3f03645995f603786b9847baf1b678b02c85757f
SHA1 (patch-contrib_scripts_vmail.cgi) = 827149400de5115eb91bdc563f2319d9bcb04cc8
SHA1 (patch-funcs_func__cdr.c) = 79c743df264948e5ea9e1c292012a1f6362d0c1e
SHA1 (patch-funcs_func__channel.c) = 9d6ed8a2431fbde6879782d8228030467aabe7eb
SHA1 (patch-funcs_func__env.c) = 9305d4dde2509f689e676295d3eb06bf5a74b3cb
SHA1 (patch-funcs_func__pjsip__aor.c) = 9874f8d66a8afd26ae1669aa727cb5fa2a788334
SHA1 (patch-funcs_func__pjsip__contact.c) = 9b1fa54ee31a549be40d487c650cc79d625c8092
SHA1 (patch-funcs_func__pjsip__endpoint.c) = 263a4bdb6365bcc9f6392d25a5aef5c607e59d04
SHA1 (patch-funcs_func__strings.c) = 08d313add57c5be822a19311fc70a7555bd63877
SHA1 (patch-include_asterisk_autoconfig.h.in) = fd4260088738c8790721a48f524b7c5d9dd10971
SHA1 (patch-include_asterisk_lock.h) = 85418bcd20f3ed7eb0310f46f3b2d334980bdcef
SHA1 (patch-include_asterisk_strings.h) = 9ace78a13131bcb411eda79a98264b5cfcc7789c
SHA1 (patch-main_Makefile) = e3b5d261fd15ffd23d81060ff3aafba6b0300e7c
SHA1 (patch-main_acl.c) = 06a9d247b19d648e9ff54ac2a234dc8ac8c023bb
SHA1 (patch-main_app.c) = 1c12bb207dcb0060017d63ba4f11fcf63d60a45e
SHA1 (patch-main_ast__expr2.c) = bad644eb956645e889344810ec315afd430853be
SHA1 (patch-main_ast__expr2.y) = 56ac74b5a3ae47bd5bec798e549ec43bd085e0e8
SHA1 (patch-main_asterisk.c) = 1262d792f330fe8a1bb1d1f7ba51bc502d65be42
SHA1 (patch-main_astmm.c) = 26a98d6fbb567ae619041ebd01a31349a847deab
SHA1 (patch-main_bridge__basic.c) = b48627e563e20544017fdfcfb4559e868badf41d
SHA1 (patch-main_bridge__channel.c) = 72dafc04521fa02e8456c09d5c9be4789d8ac918
SHA1 (patch-main_callerid.c) = 0ea1b3df8aaf3969fcd9e06055c8e6184d50d3d3
SHA1 (patch-main_cdr.c) = 540fbdb354aba100fa37392b879b92a85d1d8620
SHA1 (patch-main_cel.c) = 61f70a65f78dc21d88d32dbe67966b8f5ffb2323
SHA1 (patch-main_cli.c) = ee72bcaac7dce397354cbc09af4ed7441dbb4650
SHA1 (patch-main_conversions.c) = a516ef4f706fabbd250f66a3159825a2a6085344
SHA1 (patch-main_dns__naptr.c) = 4fa3fe5d2acf7bcd84ca2044280c644e4bd15d7f
SHA1 (patch-main_enum.c) = c5f620297cf98f95ce74aa0d98eddc697946a77b
SHA1 (patch-main_features.c) = 6e50ea4c6ee26f56edca22611aeed44787459968
SHA1 (patch-main_http.c) = b36f1f3f0da25456a17888d34ea2bf7b61c1acf4
SHA1 (patch-main_indications.c) = 511b4c270e4a4a71517109f959121777caf2aa36
SHA1 (patch-main_logger.c) = 321a52b3015af85ea13055953cec5a5d9da05ec8
SHA1 (patch-main_manager.c) = 2f88c51f4ca62985a1824efd60a39542925adf95
SHA1 (patch-main_pbx.c) = 8e7ced268edb29238f96418e8b21456364c4ae1f
SHA1 (patch-main_pbx__builtins.c) = 3e5ede8a62821fda498f2bea94af386aca01798c
SHA1 (patch-main_pbx__timing.c) = a4657330086c5b0e8fd271d5676fb897badea452
SHA1 (patch-main_sched.c) = 4219ac1561e8c4fbc5b1facdf38b3e8b764d5def
SHA1 (patch-main_stdtime_localtime.c) = c44f1f9eb142fe595f1e82167e13c456508f3879
SHA1 (patch-main_taskprocessor.c) = f90805bd78fd4096beb9ee1cf9c794c50b87481a
SHA1 (patch-main_tdd.c) = 9f525971938dd4f222622cb3e78a35822bd08389
SHA1 (patch-main_test.c) = f38b370cdb5788304e02c71ef05d2130ead9de98
SHA1 (patch-main_utils.c) = ab6fb7619111f4906ff3797e6d918dd0c8f9f7e2
SHA1 (patch-menuselect_menuselect.c) = 8bae3a2c6b8c6e7927b35bd83147a55e380efd7f
SHA1 (patch-pbx_pbx__config.c) = cc5e6d2b383f86abfb354c9bf14fc93374fba0a3
SHA1 (patch-pbx_pbx__dundi.c) = 1bc28ff2412da569f139f245c5223845a2f6cebe
SHA1 (patch-res_ael_pval.c) = 8a238c78403d3098bf8be8ae266162bc05e586f3
SHA1 (patch-res_res__calendar.c) = 45211a3baf8fbd8b201ba0167f8c56fb35728c4a
SHA1 (patch-res_res__format__attr__celt.c) = 81d5300b9a2b33e733e30760e2c9858c87b3e554
SHA1 (patch-res_res__format__attr__h263.c) = 4438d544ee028404e407d5ee3229c8f3536135f5
SHA1 (patch-res_res__format__attr__ilbc.c) = f7ff1692eae46b7950665f58317f6e39607dcc01
SHA1 (patch-res_res__format__attr__opus.c) = ba1012f111a7a996f85bbc09fec81569d2179888
SHA1 (patch-res_res__format__attr__silk.c) = d94370f9b09c917f4d68ebfbcc995c1bef1ed675
SHA1 (patch-res_res__format__attr__siren14.c) = 41e997886ca9f554e46f3af36e07e3aea984dd47
SHA1 (patch-res_res__format__attr__siren7.c) = e20e288781d0530049d127731edb8d309049077d
SHA1 (patch-res_res__format__attr__vp8.c) = 6257e281c0a29dfd3ef2613bfa5be172d399d2e0
SHA1 (patch-res_res__hep__pjsip.c) = b0c8fed52451ec31a2c77d4abd28640631bb708c
SHA1 (patch-res_res__limit.c) = e80f370fe5b84dcdc2f38e2137d5ed6f75ba35a4
SHA1 (patch-res_res__musiconhold.c) = 31f6ced9a9cfc49cbcfa68e7689a7a16586f24d9
SHA1 (patch-res_res__pjproject.c) = ed681bdf0e88c812fdc7d6c9982e82364a7ccb53
SHA1 (patch-res_res__pjsip__diversion.c) = b7996a43b4af395392161f75319ab499ceda7f09
SHA1 (patch-res_res__pjsip_pjsip__configuration.c) = 7a9f2c293ad5c8d05df5cc9b304473859ee09d6f
SHA1 (patch-res_res__xmpp.c) = 1ef89d9dab56f9c6d37dacbdd18dbd54e654050a
SHA1 (patch-sounds_Makefile) = acc15088ae2545f2822246466bfe783b5215fc54
SHA1 (patch-tests_test__locale.c) = f3f1edc86356f2a7b4d3493433c772e164c77f66
SHA1 (patch-tests_test__voicemail__api.c) = c600f726136581e47cf34da2c0bb485b8a5912eb
SHA1 (patch-third-party_pjproject_patches_0100-netbsd.patch) = fa82ca5f7340d97f9d6729734b4e698dfd26ed61
SHA1 (patch-third-party_pjproject_patches_0110-netbsd.patch) = 52e84093814dba144f89a2f9f953465f877f1506
SHA1 (patch-third-party_pjproject_patches_0120-netbsd.patch) = 1f6e9d9a1fb12dcf8efeff945a78cb3583f74598
SHA1 (patch-third-party_pjproject_patches_0130-netbsd.patch) = a1ec694ba0e2ebe1e434dc77b45ad441e730998a
SHA1 (patch-third-party_pjproject_patches_0140-netbsd.patch) = 0c18adc61339c74dfc3702b5f4428a99cb370252
SHA1 (patch-third-party_pjproject_patches_0150-netbsd.patch) = 97c6a868df8359aa27ef009863c845731a5c03a2
SHA1 (patch-third-party_pjproject_patches_0160-netbsd.patch) = d3b2aa29f368ae53951615030fb5648a12fb3426
SHA1 (patch-utils_Makefile) = 30e22c5d5d740c5531d657f91f7b51fa477d8a74
SHA1 (patch-utils_extconf.c) = f35d079c4801fe20132ff52d63d951d9e1658902
SHA1 (patch-utils_smsq.c) = 5c4cd729f1c9cb68291c514a2e54418e9b5a47cb
@


1.4
log
@Update to Asterisk 23.3.0:


## Change Log for Release asterisk-23.3.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)

### Summary:

- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0

### User Notes:

- #### acl: Add ACL support to http and ari
  A new section, type=restriction has been added to http.conf
  to allow an uri prefix based acl to be configured. See
  http.conf.sample for examples and more information.
  The user section of ari.conf can now contain an acl configuration
  to restrict users access. See ari.conf.sample for examples and more
  information

- #### res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection occurs before ICE candidate check
  WebRTC calls using TURN configured in rtp.conf (turnaddr,
  turnusername, turnpassword) will now correctly complete DTLS/SRTP
  negotiation. Previously all DTLS packets were silently dropped due to
  the loopback re-injection address not being in the ICE active candidate
  list.

- #### docs: Add "Provided-by" to doc XML and CLI output.
  The CLI help for applications, functions, manager commands and
  manager events now shows the module that provides its functionality.

- #### CDR/CEL Custom Performance Improvements
  Significant performance improvements have been made to the
  cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules.
  See the new sample config files for those modules to see how to benefit
  from them.

- #### chan_websocket: Add media direction.
  WebSocket now supports media direction, allowing for
  unidirectional media. This is done from the perspective of the
  application and can be set via channel origination, external media, or
  commands sent from the application. Check out
  https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/ for
  more.

- #### app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option
  The 'force_longest_waiting_caller' option now supports a 'prio' setting.
  When set to 'prio', calls are offered by priority first, then by wait time.

- #### Upgrade bundled pjproject to 2.16.
  Bundled pjproject has been upgraded to 2.16. For more
  information on what all is included in this change, check out the
  pjproject Github page: https://github.com/pjsip/pjproject/releases

- #### res_pjsip_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function
  A new PJSIP_HEADER option has been added that allows
  inheriting pjsip headers from the inbound to the outbound bridged
  channel.
  Example- same => n,Set(PJSIP_INHERITABLE_HEADER(add,X-custom-1)=alpha)
  will add X-custom-1: alpha to the outbound pjsip channel INVITE
  upon Dial.

- #### app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric
  Fixes an issue where QUEUE_RAISE_PENALTY=rN could raise a member’s penalty below QUEUE_MIN_PENALTY during member selection. This could allow members intended to be excluded to be selected. The queue now consistently respects the minimum penalty when raising penalties, aligning member selection behavior with queue empty checks and documented rN semantics.


## Issue and Commit Detail:

### Closed Issues:

  - 449: [bug]:  PJSIP confuses media address after INVITE requiring authentication
  - 566: [bug]: core: SIGSEGV on DTMF when no timing modules loaded
  - 1356: [bug]: MESSAGE requests should not contain a Contact header
  - 1524: [bug]: PJSIP if sdp_session is blank the initial INVITE doesn't attach an SDP offer, worked in chan_sip
  - 1611: [bug]: asterisk deadlocked on start sometimes
  - 1612: [improvement]: pjsip: Upgrade bundled version to pjproject 2.16
  - 1637: [improvement]: force_longest_waiting_caller should also consider caller priority
  - 1641: [bug]: res_pjsip_config_wizard: Endpoints fail to update when Named ACLs change after reload
  - 1651: [bug]: Asterisk crashes with munmap_chunk() when using sorcery realtime for PJSIP registration objects
  - 1657: [bug]: Wrong dtmf payload is used when inbound invite contains 8K and 16K, and outgoing leg is using G722 and SRTP
  - 1670: [new-feature]: Add new option to PJSIP_HEADER to pass headers from the inbound to outbound channel.
  - 1691: [bug]: force_longest_waiting_caller stops offering calls if a call joins at the first position
  - 1703: [bug]: res_pjsip_pubsub: ao2 reference leak of subscription tree in ast_sip_subscription
  - 1707: [bug]: chan_iax2: Crash when processing video frames with negative length
  - 1716: [bug]: Ghost call when UAC didn't respond with 487 for a cancel request from server even after original call hangup.
  - 1724: [improvement]: say.c - added language support for pashto and dari
  - 1730: [bug]: CPP channel storage get_by_name_prefix does not check prefix match
  - 1755: [bug]: app_dial, utils.h: Compilation failure with -Wold-style-declaration and -Wdiscarded-qualifiers
  - 1781: [bug]: More discarded-qualifiers errors with gcc 15.2.1
  - 1783: [bug]: Several unused-but-set-variable warnings with gcc 16
  - 1785: [bug]: chan_websocket doesn’t work with genericplc and transcoding
  - 1786: [bug]: chan_dahdi: A few more discarded-qualifiers errors not caught previously
  - 1795: [bug]: DTLS packets dropped when TURN configured in rtp.conf due to loopback re-injection occurring before ICE candidate source check
  - 1797: [bug]: Potential logic issue in translated frame write loop (main/file.c)
  - 1802: [improvement]: app_dial: Channel name should be included in warnings during wait_for_answer
  - 1804: [new-feature]: dsp.c: Add support for R2 signaling
  - 1814: [bug]: A pjsip transport with an invalid config can cause issues with other transports
  - 1816: [bug]: ARI: RTPAUDIO channel vars aren't set if call hung up by ARI.
  - 1819: [bug]: When a 302 is received from a UAS, the cause and tech_cause codes set on the channel are incorrect.
  - 1831: [bug]:raise_exception() and EXCEPTION() read use channel datastores without holding ast_channel_lock
  - 1833: [bug]: Address security vulnerabilities in pjproject
  - 1844: [bug]: cdrel_custom isn't respecting the default time format for CEL records
  - 1845: [bug]:res_cdrel_custom produces wrong float timestamps
  - 1852: [bug]: res_cdrel_custom: connection to the sqlite3 database closes from time to time

### Commit List:

-  res_cdrel_custom: do not free config when no new config was loaded
-  res_cdrel_custom: Resolve several formatting issues.
-  res_pjsip: Address pjproject security vulnerabilities
-  pbx: Hold channel lock for exception datastore access
-  xmldoc.c: Fix memory leaks in handling of provided_by.
-  SECURITY.md: Update with additional instructions.
-  res_audiosocket: Fix header read loop to use correct buffer offset
-  manager.c : Fix CLI event display
-  chan_pjsip: Set correct cause codes for non-2XX responses.
-  res_pjsip_config_wizard: Force reload on Named ACL change events
-  rtp: Set RTPAUDIOQOS variables when ast_softhangup is called.
-  channel: Prevent crash during DTMF emulation when no timing module is loaded
-  res_pjsip: Remove temp transport state when a transport fails to load.
-  res_pjsip_messaging: Remove Contact header from out-of-dialog MESSAGE as per RFC3428
-  acl: Add ACL support to http and ari
-  res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection occurs before ICE candidate check
-  dsp.c: Add support for detecting R2 signaling tones.
-  app_dial: Include channel name in warnings during wait_for_answer.
-  main/file: fix translated-frame write loop to use current frame
-  docs: Add "Provided-by" to doc XML and CLI output.
-  chan_websocket_doc.xml: Add d(media_direction) option.
-  resource_channels.c: Fix validation response for externalMedia with AudioSockets
-  CDR/CEL Custom Performance Improvements
-  chan_websocket: Remove silence generation and frame padding.
-  chan_websocket: Add media direction.
-  fix: Add macOS (Darwin) compatibility for building Asterisk
-  astconfigparser.py: Fix regex pattern error by properly escaping string
-  res_rtp_asterisk: use correct sample rate lookup to account for g722
-  res_pjsip_outbound_registration.c: Prevent crash if load_module() fails
-  pjsip_configuration: Ensure s= and o= lines in SDP are never empty
-  res_pjsip_session: Make sure NAT hook runs when packet is retransmitted for whatever reason.
-  chan_dahdi: Fix discarded-qualifiers errors.
-  build: Fix unused-but-set-variable warnings with gcc 16.
-  build: Fix another GCC discarded-qualifiers const error.
-  chan_iax2: Fix crash due to negative length frame lengths.
-  build: Fix GCC discarded-qualifiers const errors.
-  endpoints: Allow access to latest snapshot directly.
-  app_dial, utils.h: Avoid old style declaration and discarded qualifier.
-  app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option
-  Upgrade bundled pjproject to 2.16.
-  res_pjsip_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function
-  app_queue: Queue Timing Parity with Dial() and Accurate Wait Metrics
-  stasis.c: Fix deadlock in stasis_topic_pool_get_topic during module load
-  app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric
-  app_queue: Only compare calls at 1st position across queues when forcing longest waiting caller.
-  channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match
-  app_amd: Remove errant space in documentation for totalAnalysisTime.
-  say.c: added language support for pashto and dari
-  res_pjsip_session.c: Prevent INVITE failover when session is cancelled
-  res_pjsip_pubsub: Fix ao2 reference leak of subscription tree in ast_sip_subscription
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.3 2026/02/16 04:11:13 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-23.3.0/asterisk-23.3.0.tar.gz) = 6ea1459025e482aaa49fe425c262b0fa93191e03f9459add7601dcc6be8b2621
SHA512 (asterisk-23.3.0/asterisk-23.3.0.tar.gz) = 90b8d98856bc9654986b9af7e502c6c8cba49cd9a686f437563999d1d262bd57dc0fabd8fc7478370a810af31df3d123dcb5e8ebfb047bb206692f6526dc5f25
Size (asterisk-23.3.0/asterisk-23.3.0.tar.gz) = 26448744 bytes
BLAKE2s (asterisk-23.3.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-23.3.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-23.3.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-23.3.0/pjproject-2.16.md5) = 1db3bea58adb58231de14971b28b7de2691da688e928b86eb75aa1828442de7c
SHA512 (asterisk-23.3.0/pjproject-2.16.md5) = eb099d845b774dcad86a53694a488055c7bf5acb172fdbe904682a780ab46ef9369f9c9dcd462af477359aa1d72e37256be4c92be612ca7b62f03d6d0bd727c7
Size (asterisk-23.3.0/pjproject-2.16.md5) = 166 bytes
BLAKE2s (asterisk-23.3.0/pjproject-2.16.tar.bz2) = 7f24150c9b62be590aea7077ad932e0ea6e34ab755fa25a31a5b095b978ed0e8
SHA512 (asterisk-23.3.0/pjproject-2.16.tar.bz2) = a4c693f95e28d3f917adb7f27519adcd0286b39bd79e8bd39a3f84a550f99f5ad7951081f94a83a6e6fdb3acf7bfe38692b3682852a5921fe52ad134e997e820
Size (asterisk-23.3.0/pjproject-2.16.tar.bz2) = 8783258 bytes
d26 1
a26 1
SHA1 (patch-build__tools_make__xml__documentation) = f1d98a6dd2e2d4f1df202e35b2bd0f64c4af3b51
@


1.3
log
@update to Asterisk 23.2.2:

----- 23.2.2 -----

## Change Log for Release asterisk-23.2.2

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)

### 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.

----- 23.2.1 -----

## Change Log for Release asterisk-23.2.1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)

### Summary:

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

## Issue and Commit Detail:

### Closed Issues:

  - 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers when system locale is non-standard

----- 23.2.0 -----

## Change Log for Release asterisk-23.2.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)

### Summary:

- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0

### User Notes:

- #### chan_websocket.conf.sample: Fix category name.
  The category name in the chan_websocket.conf.sample file was
  incorrect.  It should be "global" instead of "general".

- #### cli.c: Allow 'channel request hangup' to accept patterns.
  The 'channel request hangup' CLI command now accepts
  multiple channel names, POSIX Extended Regular Expressions, glob-like
  patterns, or a combination of all of them. See the CLI command 'core
  show help channel request hangup' for full details.

- #### res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache populate command
  The AMI command sorcery memory cache populate will now
  return an error if there is an internal error performing the populate.
  The CLI command will display an error in this case as well.

- #### res_geolocation:  Fix multiple issues with XML generation.
  Geolocation: Two new optional profile parameters have been added.
  * `pidf_element_id` which sets the value of the `id` attribute on the top-level
    PIDF-LO `device`, `person` or `tuple` elements.
  * `device_id` which sets the content of the `<deviceID>` element.
  Both parameters can include channel variables.

- #### res_pjsip_messaging: Add support for following 3xx redirects
  A new pjsip endpoint option follow_redirect_methods was added.
  This option is a comma-delimited, case-insensitive list of SIP methods
  for which SIP 3XX redirect responses are followed. An alembic upgrade
  script has been added for adding this new option to the Asterisk
  database.

- #### taskprocessors: Improve logging and add new cli options
  New CLI command has been added -
  core show taskprocessor name <taskprocessor-name>

- #### ccss:  Add option to ccss.conf to globally disable it.
  A new "enabled" parameter has been added to ccss.conf.  It defaults
  to "yes" to preserve backwards compatibility but CCSS is rarely used so
  setting "enabled = no" in the "general" section can save some unneeded channel
  locking operations and log message spam.  Disabling ccss will also prevent
  the func_callcompletion and chan_dahdi modules from loading.

- #### Makefile: Add module-list-* targets.
  Try "make module-list-deprecated" to see what modules
  are on their way out the door.

- #### app_mixmonitor: Add 's' (skip) option to delay recording.
  This change introduces a new 's(<seconds>)' (skip) option to the MixMonitor
  application. Example:
    MixMonitor(${UNIQUEID}.wav,s(3))
  This skips recording for the first 3 seconds before writing audio to the file.
  Existing MixMonitor behavior remains unchanged when the 's' option is not used.

- #### app_queue.c: Only announce to head caller if announce_to_first_user
  When announce_to_first_user is false, no announcements are played to the head caller


### Upgrade Notes:

- #### res_geolocation:  Fix multiple issues with XML generation.
  Geolocation: In order to correct bugs in both code and
  documentation, the following changes to the parameters for GML geolocation
  locations are now in effect:
  * The documented but unimplemented `crs` (coordinate reference system) element
    has been added to the location_info parameter that indicates whether the `2d`
    or `3d` reference system is to be used. If the crs isn't valid for the shape
    specified, an error will be generated. The default depends on the shape
    specified.
  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.  If crs isn't
    specified, it will default to `2d` for these shapes.
    The Sphere, Ellipsoid and Prism shapes MUST use a `3d` crs. If crs isn't
    specified, it will default to `3d` for these shapes.
    The Point and Polygon shapes may use either crs.  The default crs is `2d`
    however so if `3d` positions are used, the crs must be explicitly set to `3d`.
  * The `geoloc show gml_shape_defs` CLI command has been updated to show which
    coordinate reference systems are valid for each shape.
  * The `pos3d` element has been removed in favor of allowing the `pos` element
    to include altitude if the crs is `3d`.  The number of values in the `pos`
    element MUST be 2 if the crs is `2d` and 3 if the crs is `3d`.  An error
    will be generated for any other combination.
  * The angle unit-of-measure for shapes that use angles should now be included
    in the respective parameter.  The default is `degrees`. There were some
    inconsistent references to `orientation_uom` in some documentation but that
    parameter never worked and is now removed.  See examples below.
  Examples...
  ```
    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
    location_info = shape="Point", pos="39.0 -105.0"
    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
    device_id = mac:001122334455
    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
  ```

- #### pjsip: Move from threadpool to taskpool
  The threadpool_* options in pjsip.conf have now
  been deprecated though they continue to be read and used.
  They have been replaced with taskpool options that give greater
  control over the underlying taskpool used for PJSIP. An alembic
  upgrade script has been added to add these options to realtime
  as well.

- #### app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
  In an effort to reduce log spam, two normal progress
  "pickup attempted" log messages from app_directed_pickup have been changed
  from NOTICE to VERBOSE(3).  This puts them on par with other normal
  dialplan progress messages.


### Developer Notes:

- #### ccss:  Add option to ccss.conf to globally disable it.
  A new API ast_is_cc_enabled() has been added.  It should be
  used to ensure that CCSS is enabled before making any other ast_cc_* calls.

- #### chan_websocket: Add ability to place a MARK in the media stream.
  Apps can now send a `MARK_MEDIA` command with an optional
  `correlation_id` parameter to chan_websocket which will be placed in the
  media frame queue. When that frame is dequeued after all intervening media
  has been played to the core, chan_websocket will send a
  `MEDIA_MARK_PROCESSED` event to the app with the same correlation_id
  (if any).

- #### chan_websocket: Add capability for JSON control messages and events.
  The chan_websocket plain-text control and event messages are now
  deprecated (but remain the default) in favor of JSON formatted messages.
  See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for
  more information.
  A "transport_data" parameter has been added to the
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.2 2025/12/01 04:26:02 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-23.2.2/asterisk-23.2.2.tar.gz) = ebad214227763b4c1bdfad3037e85ee916d015e5af0175567d7e326b62b87a55
SHA512 (asterisk-23.2.2/asterisk-23.2.2.tar.gz) = f990a51d37c96820bf1709c7fa43cfeb90b2862ddcd2075248c09c538df38ebdc5d1906d6522c2320a49a0179a26b157fa3db3b5f33731d59f0d1b73fd81f88f
Size (asterisk-23.2.2/asterisk-23.2.2.tar.gz) = 26385116 bytes
BLAKE2s (asterisk-23.2.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-23.2.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-23.2.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-23.2.2/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-23.2.2/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-23.2.2/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-23.2.2/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-23.2.2/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-23.2.2/pjproject-2.15.1.tar.bz2) = 8492214 bytes
d26 1
a26 1
SHA1 (patch-build__tools_make__xml__documentation) = 2c617cfdc96b1ddf51736205b83e1b737c110ad9
d33 1
a33 1
SHA1 (patch-configure) = a9199103edade280e0cf469360ec0d9b9e91ed1a
d43 1
a43 1
SHA1 (patch-include_asterisk_autoconfig.h.in) = 0adedab622e81b7103b61efd2adc0e8af7b4af9f
d57 1
a57 1
SHA1 (patch-main_cel.c) = 22fa21db8e0afa0958d34014f52e2c4fe9c73ba2
d71 1
a71 1
SHA1 (patch-main_stdtime_localtime.c) = 1e3c62d70eab62c46ac29e03e842229cf7587d2b
d92 1
a92 1
SHA1 (patch-res_res__pjproject.c) = 0326bf12d9f798c8eae2eff4fad8b86d4bbc0589
@


1.2
log
@Update to Asterisk 23.1.0.


## Change Log for Release asterisk-23.1.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)

### Summary:

- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0

### User Notes:

- #### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
  The STIR_SHAKEN_ATTESTATION dialplan function has been added
  which will allow suppressing attestation on a call-by-call basis
  regardless of the profile attached to the outgoing endpoint.

- #### func_channel: Allow R/W of ADSI CPE capability setting.
  CHANNEL(adsicpe) can now be read or written to change
  the channels' ADSI CPE capability setting.

- #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
  Added a new option to HANGUPCAUSE to access additional
  information about hangup reason. Reason headers from pjsip
  could be read using 'tech_extended' cause type.

- #### func_math: Add DIGIT_SUM function.
  The DIGIT_SUM function can be used to return the digit sum of
  a number.

- #### app_sf: Add post-digit timer option to ReceiveSF.
  The 't' option for ReceiveSF now allows for a timer since
  the last digit received, in addition to the number-wide timeout.

- #### app_dial: Allow fractional seconds for dial timeouts.
  The answer and progress dial timeouts now have millisecond
  precision, instead of having to be whole numbers.

- #### chan_dahdi: Add DAHDI_CHANNEL function.
  The DAHDI_CHANNEL function allows for getting/setting
  certain properties about DAHDI channels from the dialplan.


### Upgrade Notes:

- #### app_queue.c: Fix error in Queue parameter documentation.
  As part of Asterisk 21, macros were removed from Asterisk.
  This resulted in argument order changing for the Queue dialplan
  application since the macro argument was removed. Upgrade notice was
  missed when this was done, so this upgrade note has been added to
  provide a record of such and a notice to users who may have not upgraded
  yet.

- #### res_audiosocket: add message types for all slin sample rates
  New audiosocket message types 0x11 - 0x18 has been added
  for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
  slin192 audio. External applications using audiosocket may need to be
  updated to support these message types if the audiosocket channel is
  created with one of these audio formats.

- #### taskpool: Add taskpool API, switch Stasis to using it.
  The threadpool_* options in stasis.conf have now been deprecated
  though they continue to be read and used. They have been replaced with taskpool
  options that give greater control over the underlying taskpool used for stasis.


### Developer Notes:

- #### chan_pjsip: Add technology-specific off-nominal hangup cause to events.
  A "tech_cause" parameter has been added to the
  ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause"
  parameter has been added to the HangupRequest, SoftHangupRequest and Hangup
  AMI event messages.  For chan_pjsip, these will be set to the last SIP
  response status code for off-nominally terminated calls.  The parameter is
  suppressed for nominal termination.

- #### ARI: The bridges play and record APIs now handle sample rates > 8K correctly.
  The ARI /bridges/play and /bridges/record REST APIs have new
  parameters that allow the caller to specify the format to be used on the
  "Announcer" and "Recorder" channels respecitvely.

- #### taskpool: Add taskpool API, switch Stasis to using it.
  The taskpool API has been added for common usage of a
  pool of taskprocessors. It is suggested to use this API instead of the
  threadpool+taskprocessor approach.


## Issue and Commit Detail:

### Closed Issues:

  - 781: [improvement]: Allow call by call disabling Stir/Shaken header inclusion
  - 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*
  - 1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance
  - 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)
  - 1478: [improvement]: Stasis threadpool -> taskpool
  - 1479: [bug]: The ARI bridge play and record APIs limit audio bandwidth by forcing the slin8 format.
  - 1483: [improvement]: sig_analog: Eliminate possible timeout for Last Number Redial
  - 1485: [improvement]: func_scramble: Add example to XML documentation.
  - 1487: [improvement]: app_dial: Allow partial seconds to be used for dial timeouts
  - 1489: [improvement]: config_options.c: Improve misleading error message
  - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock (`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup
  - 1493: [new-feature]: app_sf: Add post-digit timer option
  - 1496: [improvement]: dsp.c: Minor fixes to debug log messages
  - 1499: [new-feature]: func_math: Add function to return the digit sum
  - 1501: [improvement]: codec_builtin: Fix some inaccurate quality weights.
  - 1505: [improvement]: res_fax: Add XML documentation for channel variables
  - 1507: [improvement]: res_tonedetect: Minor formatting issue in documentation
  - 1509: [improvement]: res_fax.c — log debug error as debug, not regular log
  - 1510: [new-feature]: sig_analog: Allow '#' to end the inter-digit timeout when dialing.
  - 1514: [improvement]: func_channel: Allow R/W of ADSI CPE capability setting.
  - 1517: [improvement]: core_unreal: Preserve ADSI capability when dialing Local channels
  - 1519: [improvement]: app_dial / func_callerid: DNIS information is not propagated by Dial
  - 1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message
  - 1534: [bug]: app_queue when using gosub breaks dialplan when going from 20 to 21, What's new in 21 doesn't mention it's a breaking change,
  - 1535: [bug]: chan_pjsip changes SSRC on WebRTC channels, which is unsupported by some browsers
  - 1536: [bug]: asterisk -rx connects to console instead of executing a command
  - 1539: [bug]: safe_asterisk without TTY doesn't log to file
  - 1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is  Details are missing
  - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361
  - 1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
  - 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.1 2025/10/27 06:58:36 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-23.1.0/asterisk-23.1.0.tar.gz) = 3351e4eb1abb03f5cde8bb7442a6128d0581674c44bbca99359d53739dac00ef
SHA512 (asterisk-23.1.0/asterisk-23.1.0.tar.gz) = 97e9a3a45a601a1085f871067b66683e0ed5d4a527994117db72242a697ca3c3d6f8f827e6d0db75b552b6822a47189a5e73b20a21dc92c549ee49bed77fedbc
Size (asterisk-23.1.0/asterisk-23.1.0.tar.gz) = 26327393 bytes
BLAKE2s (asterisk-23.1.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-23.1.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-23.1.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-23.1.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-23.1.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-23.1.0/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-23.1.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-23.1.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-23.1.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
d33 1
a33 1
SHA1 (patch-configure) = e54e099c028860092127bd20860909aa00dd426b
d43 1
a43 1
SHA1 (patch-include_asterisk_autoconfig.h.in) = 254a3bc3e8d253a9ffa3ec74cd0a29fa07316fab
@


1.1
log
@comms/asterisk23: import asterisk-23.0.0

Asterisk is a complete PBX in software.  It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).

This is a standard version.  It is scheduled to go to security
fixes only on October 15th, 2026, and EOL on October 15th, 2027.
See here for more information about Asterisk versions:
https://docs.asterisk.org/About-the-Project/Asterisk-Versions/
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.6 2025/10/27 04:35:13 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-23.0.0/asterisk-23.0.0.tar.gz) = 8fce409c1b4367d5286320b4e30b3ffe9c855260cc0ddfe4eee2a55430f6a4b8
SHA512 (asterisk-23.0.0/asterisk-23.0.0.tar.gz) = 5b7170e0868e24461d279901aceca2ea400a7c4626f509c36c51b7c7aafe3a30001bf6c67cd82cd26d6e04720752e8d15b7e47fa60d81789fa6708f2f9939195
Size (asterisk-23.0.0/asterisk-23.0.0.tar.gz) = 26284333 bytes
BLAKE2s (asterisk-23.0.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-23.0.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-23.0.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-23.0.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-23.0.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-23.0.0/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-23.0.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-23.0.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-23.0.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@