head	1.12;
access;
symbols
	pkgsrc-2026Q2:1.12.0.2
	pkgsrc-2026Q2-base:1.12
	pkgsrc-2026Q1:1.11.0.2
	pkgsrc-2026Q1-base:1.11
	pkgsrc-2025Q4:1.10.0.2
	pkgsrc-2025Q4-base:1.10
	pkgsrc-2025Q3:1.8.0.2
	pkgsrc-2025Q3-base:1.8
	pkgsrc-2025Q2:1.6.0.2
	pkgsrc-2025Q2-base:1.6
	pkgsrc-2025Q1:1.4.0.2
	pkgsrc-2025Q1-base:1.4
	pkgsrc-2024Q4:1.3.0.2
	pkgsrc-2024Q4-base:1.3
	pkgsrc-2024Q3:1.2.0.4
	pkgsrc-2024Q3-base:1.2
	pkgsrc-2024Q2:1.2.0.2
	pkgsrc-2024Q2-base:1.2;
locks; strict;
comment	@# @;


1.12
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desc
@@


1.12
log
@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
@
text
@$NetBSD: distinfo,v 1.11 2026/02/16 02:49:34 jnemeth Exp $

BLAKE2s (asterisk-21.12.2/asterisk-21.12.2.tar.gz) = ee55a88bf1c85c068dbfc4346ef2cda11cb6ecf61916e046a78cee411f4fe90f
SHA512 (asterisk-21.12.2/asterisk-21.12.2.tar.gz) = 821a78ea484fc43d2745a4e261663fcfd776d699df99bc5c995507aacab8c0f852952b217b877d9897f4308e9528d08a4009acb9717eec538ee693e9e9d8eac4
Size (asterisk-21.12.2/asterisk-21.12.2.tar.gz) = 26608590 bytes
BLAKE2s (asterisk-21.12.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.12.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.12.2/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.12.2/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.12.2/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.12.2/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.12.2/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.12.2/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.12.2/pjproject-2.15.1.tar.bz2) = 8492214 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) = 2c617cfdc96b1ddf51736205b83e1b737c110ad9
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) = a73d5466342c79be9dac3a46796684cebed5ea10
SHA1 (patch-configure.ac) = 511a3ecbbb404263d4d6c4773b0a0ad44c9adf6e
SHA1 (patch-contrib_scripts_vmail.cgi) = 7935ce96ea319eb19cc2ce999813eb837d5357c0
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) = 1ea5be5e11841700e41aa101e142b21c89916636
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) = 22fa21db8e0afa0958d34014f52e2c4fe9c73ba2
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) = 1e3c62d70eab62c46ac29e03e842229cf7587d2b
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) = 401999cefa3805f63df33424c635ad18a7d00748
SHA1 (patch-res_res__pjproject.c) = 0326bf12d9f798c8eae2eff4fad8b86d4bbc0589
SHA1 (patch-res_res__pjsip__diversion.c) = b7996a43b4af395392161f75319ab499ceda7f09
SHA1 (patch-res_res__pjsip_pjsip__configuration.c) = 7a9f2c293ad5c8d05df5cc9b304473859ee09d6f
SHA1 (patch-res_res__xmpp.c) = f8619721cf0f9d8bed08eb35f529bfaa0c1ac19c
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_db1-ast_include_db.h) = 03b43353b7967f999ace3eb160828c530e2e8fae
SHA1 (patch-utils_extconf.c) = f35d079c4801fe20132ff52d63d951d9e1658902
SHA1 (patch-utils_smsq.c) = 5c4cd729f1c9cb68291c514a2e54418e9b5a47cb
@


1.11
log
@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
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.10 2025/12/01 03:42:23 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-21.12.1/asterisk-21.12.1.tar.gz) = 9dfc85c6f103e8dc7ce4ab535d35cc1bb1707f922393fadec110fd8d3c86285e
SHA512 (asterisk-21.12.1/asterisk-21.12.1.tar.gz) = aad2072aa3ea0a1cc31f74204bf2f9a907c2c103b328cba5fb69311f213ca3ddb0862398c8a970a8702a0075b3be38c587e4f944c56aa385eb38397d57b991af
Size (asterisk-21.12.1/asterisk-21.12.1.tar.gz) = 26606158 bytes
BLAKE2s (asterisk-21.12.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.12.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.12.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.12.1/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.12.1/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.12.1/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.12.1/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.12.1/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.12.1/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.10
log
@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
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.9 2025/10/27 04:07:20 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-21.12.0/asterisk-21.12.0.tar.gz) = be63cc0ea7b06430c84ddacab68a9e9feae2d976ca898b6e8074385e1a73de14
SHA512 (asterisk-21.12.0/asterisk-21.12.0.tar.gz) = 0d8addd4b16de1b0644b89105c33807127c87e50217403bd26701ff021f47cf2b746cdb047cbb8f0ec961fb5641e9fd93340cb1422a314d0a1215f5e4c169be4
Size (asterisk-21.12.0/asterisk-21.12.0.tar.gz) = 26600343 bytes
BLAKE2s (asterisk-21.12.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.12.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.12.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.12.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.12.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.12.0/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.12.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.12.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.12.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.9
log
@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
@
text
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d3 12
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BLAKE2s (asterisk-21.11.0/asterisk-21.11.0.tar.gz) = a8d5f43ee01e3ca5f6e6a338e285645e8f4c50e4b5da994f9d8eb67a090d5618
SHA512 (asterisk-21.11.0/asterisk-21.11.0.tar.gz) = a3da502b1c2dd1dafa20d7d277586b7fc545a842e62a78732ae67d7fcfd3cd90853ff0f4adfea77d0bbd873f9d189f4cb0ad11ef5f6c542b0167da87c2cf575f
Size (asterisk-21.11.0/asterisk-21.11.0.tar.gz) = 26594761 bytes
BLAKE2s (asterisk-21.11.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.11.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.11.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.11.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.11.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.11.0/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.11.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.11.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.11.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.8
log
@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
@
text
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SHA512 (asterisk-21.10.1/asterisk-21.10.1.tar.gz) = 0b972be132f8a3ed67cb880a3255db1bfb7f9c2fc2cf51fcc1a348ff3481588a1896f4041a032dccad10bebeaebdf786ab94ee69389acbf2e7676c2224063601
Size (asterisk-21.10.1/asterisk-21.10.1.tar.gz) = 26541550 bytes
BLAKE2s (asterisk-21.10.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.10.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.10.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.10.1/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.10.1/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.10.1/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.10.1/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.10.1/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.10.1/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.7
log
@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: http://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 http://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
@
text
@d1 1
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$NetBSD: distinfo,v 1.6 2025/06/02 04:37:03 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-21.10.0/asterisk-21.10.0.tar.gz) = 06e21c3a0e2188008f99ec5f6636a850a00502235162bc1b78f8ce395ceba004
SHA512 (asterisk-21.10.0/asterisk-21.10.0.tar.gz) = 99829addcd2f09d9a7135a325252b67c64e4aefb5bfb6d24478e3c06ac5b0ada962aedbc482bde4bc790ae8eb98a4ce58fc7a8e25c5b4269dd0d76f22de5da7f
Size (asterisk-21.10.0/asterisk-21.10.0.tar.gz) = 26538178 bytes
BLAKE2s (asterisk-21.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.10.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.10.0/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.10.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.10.0/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.10.0/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.10.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.10.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.6
log
@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.
@
text
@d1 1
a1 1
$NetBSD: distinfo,v 1.5 2025/05/19 06:57:34 jnemeth Exp $
d3 12
a14 12
BLAKE2s (asterisk-21.9.1/asterisk-21.9.1.tar.gz) = c60af39bfa030808cef5752b7443d78ffc74e832929a906432b2978f44d06005
SHA512 (asterisk-21.9.1/asterisk-21.9.1.tar.gz) = 7212be039a16602a333c623b76b5f3c829bd058825eec6151b59338bc405f43ae535d92fddcf9e5f2f9f1c0745ceacabda6549580f2d265e44ce088f7fe331e1
Size (asterisk-21.9.1/asterisk-21.9.1.tar.gz) = 26493632 bytes
BLAKE2s (asterisk-21.9.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.9.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.9.1/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.9.1/pjproject-2.15.1.md5) = 1bdb00828816aff69f43eaacd084bd7d0a48670af33110bd0cd6325ead45aa48
SHA512 (asterisk-21.9.1/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
Size (asterisk-21.9.1/pjproject-2.15.1.md5) = 172 bytes
BLAKE2s (asterisk-21.9.1/pjproject-2.15.1.tar.bz2) = 2bcb38884531f0be966c78b6bac45ac63d8c612c060da91c584d192fe0fdf9cd
SHA512 (asterisk-21.9.1/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.9.1/pjproject-2.15.1.tar.bz2) = 8492214 bytes
@


1.5
log
@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
@
text
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SHA512 (asterisk-21.9.0/asterisk-21.9.0.tar.gz) = ec9659589897361cfd4c4b8d55c197a6c0b06fe1c2afbf7687a098b04265bc88d9a4f4df08676ef0bc364e7629e0096e528e78a3967510a7ab22c7fdfdcb62b1
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Size (asterisk-21.9.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
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SHA512 (asterisk-21.9.0/pjproject-2.15.1.md5) = 75963b64e702a5810fd5b8b574a07396cab1a87543d806135e7a9b9762d35129354f99283252f40ad75a6a94cd1921f164ed8e63174de0c5430e5c6913d21744
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SHA512 (asterisk-21.9.0/pjproject-2.15.1.tar.bz2) = c080eb44b49fccadb1c76ff2b3221935b0d531a1e2087b47c21b4ec2cdd8ee0e62b13c334495c9c759b348a0792204611987089a6aa6264999f0116aec8dbdfd
Size (asterisk-21.9.0/pjproject-2.15.1.tar.bz2) = 8492214 bytes
d33 2
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SHA1 (patch-configure) = 03e0de2aef9ba3143c0c457d9ec658483a2570ab
SHA1 (patch-configure.ac) = b972730a2be3bf54502116f1f7e03afee76a02cc
@


1.4
log
@asterisk[19,21,22]: Fix invalid XML documentation building
@
text
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SHA512 (asterisk-21.5.0/asterisk-21.5.0.tar.gz) = 4c8200d1e5eba1a3005dc9709be5893ef395c7635df9e64769f4e30c39b8b82be4332a829c0516bd22748f37f5be506d8f3f886381d7d0ea772d0648166c4942
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SHA512 (asterisk-21.5.0/pjproject-2.14.1.md5) = 25ce388adcd7eaa2c21d95a58d9fc5e33a6cb96dd99c292574b8f11f6f1f985cf91f91ea252300bd1be192e396ac6c8a35a87b219864339798bf6195a7650c00
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SHA512 (asterisk-21.5.0/pjproject-2.14.1.tar.bz2) = 996116df4a18fb28c8f68d122466f8664958226a38e432b6190b92fbf277b278d370a4b44fabeaf25691e3cdcde28a8879b2738ead5387d119229db01ce121d8
Size (asterisk-21.5.0/pjproject-2.14.1.tar.bz2) = 8379251 bytes
d33 1
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SHA1 (patch-configure) = 7bb72c26abe5c362bf8e415821534b83f6241473
d43 1
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SHA1 (patch-include_asterisk_autoconfig.h.in) = 23807b08b94f5cf9c2de76c2928f7ae38997d006
a58 1
SHA1 (patch-main_config.c) = 0647c59c4be846e7a9f6d523fbc93c54dc45b664
d95 1
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SHA1 (patch-res_res__xmpp.c) = 390376180d1fb11a41c16f59dd44f506006a8e5d
@


1.3
log
@Upgrade to Asterisk 21.5.0.


## Change Log for Release asterisk-21.5.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip_notify: add dialplan application
  A new dialplan application PJSIPNotify is now available
  which can send SIP NOTIFY requests from the dialplan.
  The pjsip send notify CLI command has also been enhanced to allow
  sending NOTIFY messages to a specific channel. Syntax:
  pjsip send notify <option> channel <channel>

- #### channel: Add multi-tenant identifier.
  tenantid has been added to channels. It can be read in
  dialplan via CHANNEL(tenantid), and it can be set using
  Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
  use the new tenantid option for pjsip endpoints (e.g., tenantid=My
  tenant ID) so that it will show up in Newchannel events. You can set it
  like any other channel variable using set_var in pjsip.conf as well, but
  note that this will NOT show up in Newchannel events. Tenant ID is also
  available in CDR and can be accessed with CDR(tenantid). The peer tenant
  ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
  as well if it has been set.

- #### res_pjsip_config_wizard.c: Refactor load process
  The res_pjsip_config_wizard.so module can now be reloaded.


### Upgrade Notes:

- #### channel: Add multi-tenant identifier.
  A new versioned struct (ast_channel_initializers) has been
  added that gets passed to __ast_channel_alloc_ap. The new function
  ast_channel_alloc_with_initializers should be used when creating
  channels that require the use of this struct. Currently the only value
  in the struct is for tenantid, but now more fields can be added to the
  struct as necessary rather than the __ast_channel_alloc_ap function. A
  new option (tenantid) has been added to endpoints in pjsip.conf as well.
  CEL has had its version bumped to include tenant ID.


### Commit Authors:

- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)

## Issue and Commit Detail:

### Closed Issues:

  - 740: [new-feature]: Add multi-tenant identifier to chan_pjsip
  - 763: [bug]: autoservice thread stuck in an endless sleep
  - 780: [bug]: Infinite loop of "Indicated Video Update", max CPU usage
  - 799: [improvement]: Add PJSIPNOTIFY dialplan application
  - 801: [bug]: res_stasis: Occasional 200ms delay adding channel to a bridge
  - 809: [bug]: CLI stir_shaken show verification kills asterisk
  - 816: [bug]: res_pjsip_config_wizard doesn't load properly if res_pjsip is loaded first
  - 845: [bug]: Buffer overflow in handling of security mechanisms in res_pjsip
  - 847: [bug]: Asterisk not using negotiated fall-back 8K digits
  - 854: [bug]:  wrong properties in stir_shaken.conf.sample
  - 856: [bug]: res_pjsip_sdp_rtp leaks astobj2 ast_format
  - 861: [bug]: ChanSpy unable to read audiohook read direction frame when no packet lost on both side of the call
  - 876: [bug]: ChanSpy unable to write whisper_audiohook when set flag OPTION_READONLY
  - 879: [bug]: res_stir_shaken/verification.c: Getting verification errors when global_disable=yes
  - 884: [bug]: A ':' at the top of in stir_shaken.conf make Asterisk producing a core file when starting
  - 889: [bug]: res_stir_shaken/verification.c has a stale include for jansson.h that can cause compilation to fail
  - 904: [bug]: stir_shaken: attest_level isn't being propagated correctly from attestation to profile to tn

### Commits By Author:

- #### Alexei Gradinari (2):
  - res_pjsip_sdp_rtp fix leaking astobj2 ast_format
  - autoservice: Do not sleep if autoservice_stop is called within autoservice thr..

- #### Ben Ford (1):
  - channel: Add multi-tenant identifier.

- #### Cade Parker (1):
  - chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay

- #### George Joseph (11):
  - bridge_softmix: Fix queueing VIDUPDATE control frames
  - res_pjsip_config_wizard.c: Refactor load process
  - stir_shaken: CRL fixes and a new CLI command
  - manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
  - stir_shaken.conf.sample: Fix bad references to private_key_path
  - security_agreements.c: Refactor the to_str functions and fix a few other bugs
  - app_voicemail: Use ast_asprintf to create mailbox SQL query
  - res_stir_shaken: Check for disabled before param validation
  - res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
  - res_stir_shaken: Remove stale include for jansson.h in verification.c
  - stir_shaken: Fix propagation of attest_level and a few other values

- #### Jaco Kroon (1):
  - configure:  Use . file rather than source file.

- #### Mike Bradeen (3):
  - res_stasis: fix intermittent delays on adding channel to bridge
  - res_pjsip_notify: add dialplan application
  - res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch

- #### Sean Bright (2):
  - alembic: Make 'revises' header comment match reality.
  - res_pjsip_logger.c: Fix 'OPTIONS' tab completion.

- #### Tinet-mucw (3):
  - res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
  - app_chanspy.c: resolving the issue with audiohook direction read
  - app_chanspy.c: resolving the issue writing frame to whisper audiohook.


### Commit List:

-  stir_shaken: Fix propagation of attest_level and a few other values
-  res_stir_shaken: Remove stale include for jansson.h in verification.c
-  res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
-  res_stir_shaken: Check for disabled before param validation
-  app_chanspy.c: resolving the issue writing frame to whisper audiohook.
-  app_voicemail: Use ast_asprintf to create mailbox SQL query
-  res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
-  app_chanspy.c: resolving the issue with audiohook direction read
-  security_agreements.c: Refactor the to_str functions and fix a few other bugs
-  res_pjsip_sdp_rtp fix leaking astobj2 ast_format
-  stir_shaken.conf.sample: Fix bad references to private_key_path
-  res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
-  alembic: Make 'revises' header comment match reality.
-  chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay
-  res_pjsip_notify: add dialplan application
-  manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
-  channel: Add multi-tenant identifier.
-  configure:  Use . file rather than source file.
-  res_stasis: fix intermittent delays on adding channel to bridge
-  res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
-  stir_shaken: CRL fixes and a new CLI command
-  res_pjsip_config_wizard.c: Refactor load process
-  bridge_softmix: Fix queueing VIDUPDATE control frames

### Commit Details:

#### stir_shaken: Fix propagation of attest_level and a few other values
  Author: George Joseph
  Date:   2024-09-24

  attest_level, send_mky and check_tn_cert_public_url weren't
  propagating correctly from the attestation object to the profile
  and tn.

  * In the case of attest_level, the enum needed to be changed
  so the "0" value (the default) was "NOT_SET" instead of "A".  This
  now allows the merging of the attestation object, profile and tn
  to detect when a value isn't set and use the higher level value.

  * For send_mky and check_tn_cert_public_url, the tn default was
  forced to "NO" which always overrode the profile and attestation
  objects.  Their defaults are now "NOT_SET" so the propagation
  happens correctly.

  * Just to remove some redundant code in tn_config.c, a bunch of calls to
  generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
  replaced with a single call to generate_acfg_common_sorcery_handlers().

  Resolves: #904

#### res_stir_shaken: Remove stale include for jansson.h in verification.c
  Author: George Joseph
  Date:   2024-09-17

  verification.c had an include for jansson.h left over from previous
  versions of the module.  Since res_stir_shaken no longer has a
  dependency on jansson, the bundled version wasn't added to GCC's
  include path so if you didn't also have a jansson development package
  installed, the compile would fail.  Removing the stale include
  was the only thing needed.

  Resolves: #889

#### res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
  Author: George Joseph
  Date:   2024-09-13

  * If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
  check_for_old_config() now returns LOAD_DECLINE instead of continuing
  on with a bad pointer.

  * If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
  assumes the config is being loaded from realtime and now returns
  LOAD_SUCCESS.  If it's actually not being loaded from realtime,
  sorcery will catch that later on.

  * Also refactored the error handling in load_module() a bit.

  Resolves: #884

#### res_stir_shaken: Check for disabled before param validation
  Author: George Joseph
  Date:   2024-09-11

  For both attestation and verification, we now check whether they've
  been disabled either globally or by the profile before validating
  things like callerid, orig_tn, dest_tn, etc.  This prevents useless
  error messages.

  Resolves: #879

#### app_chanspy.c: resolving the issue writing frame to whisper audiohook.
  Author: Tinet-mucw
  Date:   2024-09-10

  ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.

  Resolves: #876

#### autoservice: Do not sleep if autoservice_stop is called within autoservice thr..
  Author: Alexei Gradinari
  Date:   2024-09-04

  It's possible that ast_autoservice_stop is called within the autoservice thread.
  In this case the autoservice thread is stuck in an endless sleep.

  To avoid endless sleep ast_autoservice_stop must check that it's not called
  within the autoservice thread.

  Fixes: #763

#### app_voicemail: Use ast_asprintf to create mailbox SQL query
  Author: George Joseph
  Date:   2024-09-03

  ...instead of trying to calculate the length of the buffer needed
  manually.


#### res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
  Author: Mike Bradeen
  Date:   2024-08-21

  When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
  matching 4733 offers, Bob may want to use the 48K audio codec but can not
  accept 48K digits and so negotiates for a mixed set.

  Asterisk will now check Bob's offer to make sure Bob has indicated this is
  acceptible and if not, will use Bob's preference.

  Fixes: #847

#### app_chanspy.c: resolving the issue with audiohook direction read
  Author: Tinet-mucw
  Date:   2024-08-30

  ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.

  Resolves: #861

#### security_agreements.c: Refactor the to_str functions and fix a few other bugs
  Author: George Joseph
  Date:   2024-08-17

  * A static array of security mechanism type names was created.

  * ast_sip_str_to_security_mechanism_type() was refactored to do
    a lookup in the new array instead of using fixed "if/else if"
    statments.

  * security_mechanism_to_str() and ast_sip_security_mechanisms_to_str()
    were refactored to use ast_str instead of a fixed length buffer
    to store the result.

  * ast_sip_security_mechanism_type_to_str was removed in favor of
    just referencing the new type name array.  Despite starting with
    "ast_sip_", it was a static function so removing it doesn't affect
    ABI.

  * Speaking of "ast_sip_", several other static functions that
    started with "ast_sip_" were renamed to avoid confusion about
    their public availability.

  * A few VECTOR free loops were replaced with AST_VECTOR_RESET().

  * Fixed a meomry leak in pjsip_configuration.c endpoint_destructor
    caused by not calling ast_sip_security_mechanisms_vector_destroy().

  * Fixed a memory leak in res_pjsip_outbound_registration.c
    add_security_headers() caused by not specifying OBJ_NODATA in
    an ao2_callback.

  * Fixed a few ao2_callback return code misuses.

  Resolves: #845

#### res_pjsip_sdp_rtp fix leaking astobj2 ast_format
  Author: Alexei Gradinari
  Date:   2024-08-23

  PR #700 added a preferred_format for the struct ast_rtp_codecs,
  but when set the preferred_format it leaks an astobj2 ast_format.
  In the next code
  ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
  both functions ast_rtp_codecs_set_preferred_format
  and ast_format_cap_get_format increases the ao2 reference count.

  Fixes: #856

#### stir_shaken.conf.sample: Fix bad references to private_key_path
  Author: George Joseph
  Date:   2024-08-22

  They should be private_key_file.

  Resolves: #854

#### res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
  Author: Sean Bright
  Date:   2024-08-19

  Fixes #843


#### alembic: Make 'revises' header comment match reality.
  Author: Sean Bright
  Date:   2024-08-17


#### chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay
  Author: Cade Parker
  Date:   2024-08-07

  On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)


#### res_pjsip_notify: add dialplan application
  Author: Mike Bradeen
  Date:   2024-07-09

  Add dialplan application PJSIPNOTIFY to send either pre-configured
  NOTIFY messages from pjsip_notify.conf or with headers defined in
  dialplan.

  Also adds the ability to send pre-configured NOTIFY commands to a
  channel via the CLI.

  Resolves: #799

  UserNote: A new dialplan application PJSIPNotify is now available
  which can send SIP NOTIFY requests from the dialplan.

  The pjsip send notify CLI command has also been enhanced to allow
  sending NOTIFY messages to a specific channel. Syntax:

  pjsip send notify <option> channel <channel>


#### manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
  Author: George Joseph
  Date:   2024-08-08

  If you run an AMI CoreShowChannelMap on a channel that isn't in a
  bridge and you're in DEVMODE, you can get a FRACK because the
  bridge id is empty.  We now simply return an empty list for that
  request.


#### channel: Add multi-tenant identifier.
  Author: Ben Ford
  Date:   2024-05-21

  This patch introduces a new identifier for channels: tenantid. It's
  a stringfield on the channel that can be used for general purposes. It
  will be inherited by other channels the same way that linkedid is.

  You can set tenantid in a few ways. The first is to set it in the
  dialplan with the Set and CHANNEL functions:

  exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)

  It can also be accessed via CHANNEL:

  exten => example,2,NoOp(CHANNEL(tenantid))

  Another method is to use the new tenantid option for pjsip endpoints in
  pjsip.conf:

  [my_endpoint]
  type=endpoint
  tenantid=My tenant ID

  This is considered the best approach since you will be able to see the
  tenant ID as early as the Newchannel event.

  It can also be set using set_var in pjsip.conf on the endpoint like
  setting other channel variable:

  set_var=CHANNEL(tenantid)=My tenant ID

  Note that set_var will not show tenant ID on the Newchannel event,
  however.

  Tenant ID has also been added to CDR. It's read-only and can be accessed
  via CDR(tenantid). You can also get the tenant ID of the last channel
  communicated with via CDR(peertenantid).

  Tenant ID will also show up in CEL records if it has been set, and the
  version number has been bumped accordingly.

  Fixes: #740

  UserNote: tenantid has been added to channels. It can be read in
  dialplan via CHANNEL(tenantid), and it can be set using
  Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
  use the new tenantid option for pjsip endpoints (e.g., tenantid=My
  tenant ID) so that it will show up in Newchannel events. You can set it
  like any other channel variable using set_var in pjsip.conf as well, but
  note that this will NOT show up in Newchannel events. Tenant ID is also
  available in CDR and can be accessed with CDR(tenantid). The peer tenant
  ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
  as well if it has been set.

  UpgradeNote: A new versioned struct (ast_channel_initializers) has been
  added that gets passed to __ast_channel_alloc_ap. The new function
  ast_channel_alloc_with_initializers should be used when creating
  channels that require the use of this struct. Currently the only value
  in the struct is for tenantid, but now more fields can be added to the
  struct as necessary rather than the __ast_channel_alloc_ap function. A
  new option (tenantid) has been added to endpoints in pjsip.conf as well.
  CEL has had its version bumped to include tenant ID.


#### configure:  Use . file rather than source file.
  Author: Jaco Kroon
  Date:   2024-08-05

  source is a bash concept, so when /bin/sh points to another shell the
  existing construct won't work.

  Reference: https://bugs.gentoo.org/927055
  Signed-off-by: Jaco Kroon <jaco@@uls.co.za>

#### res_stasis: fix intermittent delays on adding channel to bridge
  Author: Mike Bradeen
  Date:   2024-07-10

  Previously, on command execution, the control thread was awoken by
  sending a SIGURG. It was found that this still resulted in some
  instances where the thread was not immediately awoken.

  This change instead sends a null frame to awaken the control thread,
  which awakens the thread more consistently.

  Resolves: #801

#### res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
  Author: Tinet-mucw
  Date:   2024-08-02

  When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.

  Resolves: asterisk#826

#### stir_shaken: CRL fixes and a new CLI command
  Author: George Joseph
  Date:   2024-07-19

  * Fixed a bug in crypto_show_cli_store that was causing asterisk
  to crash if there were certificate revocation lists in the
  verification certificate store.  We're also now prefixing
  certificates with "Cert:" and CRLs with "CRL:" to distinguish them
  in the list.

  * Added 'untrusted_cert_file' and 'untrusted_cert_path' options
  to both verification and profile objects.  If you have CRLs that
  are signed by a different CA than the incoming X5U certificate
  (indirect CRL), you'll need to provide the certificate of the
  CRL signer here.  Thse will show up as 'Untrusted" when showing
  the verification or profile objects.

  * Fixed loading of crl_path.  The OpenSSL API we were using to
  load CRLs won't actually load them from a directory, only a file.
  We now scan the directory ourselves and load the files one-by-one.

  * Fixed the verification flags being set on the certificate store.
    - Removed the CRL_CHECK_ALL flag as this was causing all certificates
      to be checked for CRL extensions and failing to verify the cert if
      there was none.  This basically caused all certs to fail when a CRL
      was provided via crl_file or crl_path.
    - Added the EXTENDED_CRL_SUPPORT flag as it is required to handle
      indirect CRLs.

  * Added a new CLI command...
  `stir_shaken verify certificate_file <certificate_file> [ <profile> ]`
  which will assist troubleshooting certificate problems by allowing
  the user to manually verify a certificate file against either the
  global verification certificate store or the store for a specific
  profile.

  * Updated the XML documentation and the sample config file.

  Resolves: #809

#### res_pjsip_config_wizard.c: Refactor load process
  Author: George Joseph
  Date:   2024-07-23

  The way we have been initializing the config wizard prevented it
  from registering its objects if res_pjsip happened to load
  before it.

  * We now use the object_type_registered sorcery observer to kick
  things off instead of the wizard_mapped observer.

  * The load_module function now checks if res_pjsip has been loaded
  already and if it was it fires the proper observers so the objects
  load correctly.

  Resolves: #816

  UserNote: The res_pjsip_config_wizard.so module can now be reloaded.

#### bridge_softmix: Fix queueing VIDUPDATE control frames
  Author: George Joseph
  Date:   2024-07-17

  softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else()
  with the bridge_channel so the VIDUPDATE control frame isn't echoed back.

  softmix_bridge_write_control() was setting bridge_channel to NULL
  when calling ast_bridge_queue_everyone_else() for VIDUPDATE control
  frames.  This was causing the frame to be echoed back to the
  channel it came from.  In certain cases, like when two channels or
  bridges are being recorded, this can cause a ping-pong effect that
  floods the system with VIDUPDATE control frames.

  Resolves: #780


## Change Log for Release asterisk-21.4.3

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9): A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used

### Commits By Author:

- #### George Joseph (1):
  - res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback


### Commit List:

-  res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback

### Commit Details:

#### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  Author: George Joseph
  Date:   2024-08-12

  The ub_result pointer passed to unbound_resolver_callback by
  libunbound can be NULL if the query was for something malformed
  like `.1` or `[.1]`.  If it is, we now set a 'ns_r_formerr' result
  and return instead of crashing with a SEGV.  This causes pjproject
  to simply cancel the transaction with a "No answer record in the DNS
  response" error.  The existing "off nominal" unit test was also
  updated to check this condition.

  Although not necessary for this fix, we also made
  ast_dns_resolver_completed() tolerant of a NULL result.

  Resolves: GHSA-v428-g3cw-7hv9


## Change Log for Release asterisk-21.4.2

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44): Write=originate, is sufficient permissions for code execution / System() dialplan

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code execution / System() dialplan

### Commits By Author:

- #### George Joseph (1):
  - manager.c: Add entries to Originate blacklist


### Commit List:

-  manager.c: Add entries to Originate blacklist

### Commit Details:

#### manager.c: Add entries to Originate blacklist
  Author: George Joseph
  Date:   2024-07-22

  Added Reload and DBdeltree to the list of dialplan application that
  can't be executed via the Originate manager action without also
  having write SYSTEM permissions.

  Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan
  functions that can't be executed via the Originate manager action
  without also having write SYSTEM permissions.

  If the Queue application is attempted to be run by the Originate
  manager action and an AGI parameter is specified in the app data,
  it'll be rejected unless the manager user has either the AGI or
  SYSTEM permissions.

  Resolves: #GHSA-c4cg-9275-6w44


## Change Log for Release asterisk-21.4.1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (2)

## Issue and Commit Detail:

### Closed Issues:

  - 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using "make samples"
  - 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -> 20.9.0

### Commits By Author:

- #### George Joseph (2):
  - voicemail.conf.sample: Fix ':' comment typo
  - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()


### Commit List:

-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo

### Commit Details:

#### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  Author: George Joseph
  Date:   2024-07-25

  There can be empty slots in payload_mapping_tx corresponding to
  dynamic payload types that haven't been seen before so we now
  check for NULL before attempting to use 'type' in the call to
  ast_format_cmp.

  Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()

  Resolves: #822

#### voicemail.conf.sample: Fix ':' comment typo
  Author: George Joseph
  Date:   2024-07-24

  ...and removed an errant trailing space.

  Resolves: #819


## Change Log for Release asterisk-21.4.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0

### User Notes:

- #### app_voicemail_odbc: Allow audio to be kept on disk
  This commit adds a new voicemail.conf option
  'odbc_audio_on_disk' which when set causes the ODBC variant of
  app_voicemail_odbc to leave the message and greeting audio files
  on disk and only store the message metadata in the database.
  Much more information can be found in the voicemail.conf.sample
  file.

- #### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
  will be stored in the queue log.
  If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.

- #### pbx.c: expand fields width of "core show hints"
  The fields width of "core show hints" were increased.
  The width of "extension" field to 30 characters and
  the width of the "device state id" field to 60 characters.

- #### rtp_engine: add support for multirate RFC2833 digits
  No change in configuration is required in order to enable this
  feature. Endpoints configured to use RFC2833 will automatically have this
  enabled. If the endpoint does not support this, it should not include it in
  the SDP offer/response.
  Resolves: #699


### Upgrade Notes:

- #### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Add a new column to the queues table:
  queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
  to control whether the Restricted Caller ID will be stored in the queue log.


### Commit Authors:

- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)

## Issue and Commit Detail:

### Closed Issues:

  - 699: [improvement]: Add support for multi-rate DTMF
  - 736: [bug]: Seg fault on CLI after PostgreSQL CDR module fails to load for a second time
  - 765: [improvement]: Add option to not log Restricted Caller ID to queue_log
  - 770: [improvement]: pbx.c: expand fields width of "core show hints"
  - 776: [bug] DTMF broken after rtp_engine: add support for multirate RFC2833 digits commit
  - 783: [bug]: Under certain circumstances a channel snapshot can get orphaned in the cache
  - 789: [bug]: Mediasec headers aren't sent on outgoing INVITEs
  - 797: [bug]:

### Commits By Author:

- ### Alexei Gradinari (2):
  - pbx.c: expand fields width of "core show hints"
  - app_queue:  Add option to not log Restricted Caller ID to queue_log

- ### Bastian Triller (1):
  - cli: Show configured cache dir

- ### George Joseph (4):
  - app_voicemail_odbc: Allow audio to be kept on disk
  - stasis_channels: Use uniqueid and name to delete old snapshots
  - security_agreement.c: Always add the Require and Proxy-Require headers
  - ast-db-manage: Remove duplicate enum creation

- ### Igor Goncharovsky (1):
  - res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()

- ### Mike Bradeen (2):
  - rtp_engine: add support for multirate RFC2833 digits
  - res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits

- ### Sean Bright (7):
  - file.h: Rename function argument to avoid C++ keyword clash.
  - bundled_pjproject: Disable UPnP support.
  - asterisk.c: Don't log an error if .asterisk_history does not exist.
  - xml.c: Update deprecated libxml2 API usage.
  - manager.c: Properly terminate `CoreShowChannelMap` event.
  - pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
  - logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.

- ### Tinet-mucw (1):
  - bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..

- ### Walter Doekes (1):
  - chan_ooh323: Fix R/0 typo in docs

- ### chrsmj (1):
  - cdr_pgsql: Fix crash when the module fails to load multiple times.


### Commit List:

-  res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
-  res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
-  ast-db-manage: Remove duplicate enum creation
-  security_agreement.c: Always add the Require and Proxy-Require headers
-  logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
-  stasis_channels: Use uniqueid and name to delete old snapshots
-  app_voicemail_odbc: Allow audio to be kept on disk
-  app_queue:  Add option to not log Restricted Caller ID to queue_log
-  pbx.c: expand fields width of "core show hints"
-  pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
-  manager.c: Properly terminate `CoreShowChannelMap` event.
-  cli: Show configured cache dir
-  xml.c: Update deprecated libxml2 API usage.
-  cdr_pgsql: Fix crash when the module fails to load multiple times.
-  asterisk.c: Don't log an error if .asterisk_history does not exist.
-  chan_ooh323: Fix R/0 typo in docs
-  bundled_pjproject: Disable UPnP support.
-  file.h: Rename function argument to avoid C++ keyword clash.
-  rtp_engine: add support for multirate RFC2833 digits

### Commit Details:

#### res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
  Author: Igor Goncharovsky
  Date:   2024-05-12

  When using the PJSIP_DIAL_CONTACTS() function for use in the Dial()
  command, the contacts are returned in text form, so the input to
  the path_outgoing_request() function is a contact value of NULL.
  The issue was reported in ASTERISK-28211, but was not actually fixed
  in ASTERISK-30100. This fix brings back the code that was previously
  removed and adds code to search for a contact to extract the path
  value from it.


#### res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
  Author: Mike Bradeen
  Date:   2024-06-21

  After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
  Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
  the negotiated codec(s).

  This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
  offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.

  A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
  be re-written to allow for these scenarios.

  Fixes: #776

#### ast-db-manage: Remove duplicate enum creation
  Author: George Joseph
  Date:   2024-07-08

  Remove duplicate creation of ast_bool_values from
  2b7c507d7d12_add_queue_log_option_log_restricted_.py.  This was
  causing alembic upgrades to fail since the enum was already created
  in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.

  Resolves: #797

#### security_agreement.c: Always add the Require and Proxy-Require headers
  Author: George Joseph
  Date:   2024-07-03

  The `Require: mediasec` and `Proxy-Require: mediasec` headers need
  to be sent whenever we send `Security-Client` or `Security-Verify`
  headers but the logic to do that was only in add_security_headers()
  in res_pjsip_outbound_register.  So while we were sending them on
  REGISTER requests, we weren't sending them on INVITE requests.

  This commit moves the logic to send the two headers out of
  res_pjsip_outbound_register:add_security_headers() and into
  security_agreement:ast_sip_add_security_headers().  This way
  they're always sent when we send `Security-Client` or
  `Security-Verify`.

  Resolves: #789

#### logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
  Author: Sean Bright
  Date:   2024-06-29

  Fixes #785


#### stasis_channels: Use uniqueid and name to delete old snapshots
  Author: George Joseph
  Date:   2024-05-08

  Whenver a new channel snapshot is created or when a channel is
  destroyed, we need to delete any existing channel snapshot from
  the snapshot cache.  Historically, we used the channel->snapshot
  pointer to delete any existing snapshots but this has two issues.

  First, if something (possibly ast_channel_internal_swap_snapshots)
  sets channel->snapshot to NULL while there's still a snapshot in
  the cache, we wouldn't be able to delete it and it would be orphaned
  when the channel is destroyed.  Since we use the cache to list
  channels from the CLI, AMI and ARI, it would appear as though the
  channel was still there when it wasn't.

  Second, since there are actually two caches, one indexed by the
  channel's uniqueid, and another indexed by the channel's name,
  deleting from the caches by pointer requires a sequential search of
  all of the hash table buckets in BOTH caches to find the matching
  snapshots.  Not very efficient.

  So, we now delete from the caches using the channel's uniqueid
  and name.  This solves both issues.

  This doesn't address how channel->snapshot might have been set
  to NULL in the first place because although we have concrete
  evidence that it's happening, we haven't been able to reproduce it.

  Resolves: #783

#### app_voicemail_odbc: Allow audio to be kept on disk
  Author: George Joseph
  Date:   2024-04-09

  This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
  which when set causes the ODBC variant of app_voicemail to leave
  the message and greeting audio files on disk and only store the
  message metadata in the database.  This option came from a concern
  that the database could grow to large and cause remote access
  and/or replication to become slow.  In a clustering situation
  with this option, all asterisk instances would share the same
  database for the metadata and either use a shared filesystem
  or other filesystem replication service much more suitable
  for synchronizing files.

  The changes to app_voicemail to implement this feature were actually
  quite small but due to the complexity of the module, the actual
  source code changes were greater.  They fall into the following
  categories:

  * Tracing.  The module is so complex that it was impossible to
  figure out the path taken for various scenarios without the addition
  of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
  code that's not related to the functional change.  Making this worse
  was the fact that many "if" statements in this module didn't use
  braces.  Since the tracing macros add multiple statements, many "if"
  statements had to be converted to use braces.

  * Excessive use of PATH_MAX.  Previous maintainers of this module
  used PATH_MAX to allocate character arrays for filesystem paths
  and SQL statements as though they cost nothing.  In fact, PATH_MAX
  is defined as 4096 bytes!  Some functions had (and still have)
  multiples of these.  One function has 7.  Given that the vast
  majority of installations use the default spool directory path
  `/var/spool/asterisk/voicemail`, the actual path length is usually
  less than 80 bytes.  That's over 4000 bytes wasted.  It was the
  same for SQL statement buffers.  A 4K buffer for statement that
  only needed 60 bytes.  All of these PATH_MAX allocations in the
  ODBC related code were changed to dynamically allocated buffers.
  The rest will have to be addressed separately.

  * Bug fixes.  During the development of this feature, several
  pre-existing ODBC related bugs were discovered and fixed.  They
  had to do with leaving orphaned files on disk, not preserving
  original message ids when moving messages between folders,
  not honoring the "formats" config parameter in certain circumstances,
  etc.

  UserNote: This commit adds a new voicemail.conf option
  'odbc_audio_on_disk' which when set causes the ODBC variant of
  app_voicemail_odbc to leave the message and greeting audio files
  on disk and only store the message metadata in the database.
  Much more information can be found in the voicemail.conf.sample
  file.


#### bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..
  Author: Tinet-mucw
  Date:   2024-06-13

  Resolves: https://github.com/asterisk/asterisk/issues/768

#### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Author: Alexei Gradinari
  Date:   2024-06-12

  Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event
  in the queue log if the Caller ID is restricted.

  Resolves: #765

  UpgradeNote: Add a new column to the queues table:
  queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
  to control whether the Restricted Caller ID will be stored in the queue log.

  UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
  will be stored in the queue log.
  If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.


#### pbx.c: expand fields width of "core show hints"
  Author: Alexei Gradinari
  Date:   2024-06-13

  The current width for "extension" is 20 and "device state id" is 20, which is too small.
  The "extension" field contains "ext"@@"context", so 20 characters is not enough.
  The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.

  Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.

  Resolves: #770

  UserNote: The fields width of "core show hints" were increased.
  The width of "extension" field to 30 characters and
  the width of the "device state id" field to 60 characters.


#### pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
  Author: Sean Bright
  Date:   2024-06-02

  Various SIP headers permit a URI to be prefaced with a `display-name`
  production that can include characters (like commas and parentheses)
  that are problematic for Asterisk's dialplan parser and, specifically
  in the case of this patch, the PJSIP_PARSE_URI function.

  This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
  behaves identically to `PJSIP_PARSE_URI` except that the first
  argument is now a variable name and not a literal URI.

  Fixes #756


#### manager.c: Properly terminate `CoreShowChannelMap` event.
  Author: Sean Bright
  Date:   2024-06-10

  Fixes #761


#### cli: Show configured cache dir
  Author: Bastian Triller
  Date:   2024-06-07

  Since Asterisk 19 it is possible to cache recorded files into another
  directory [1] [2].
  Show configured location of cache dir in CLI's core show settings.

  [1] ASTERISK-29143
  [2] b08427134fd51bb549f198e9f60685f2680c68d7


#### xml.c: Update deprecated libxml2 API usage.
  Author: Sean Bright
  Date:   2024-05-23

  Two functions are deprecated as of libxml2 2.12:

    * xmlSubstituteEntitiesDefault
    * xmlParseMemory

  So we update those with supported API.

  Additionally, `res_calendar_caldav` has been updated to use libxml2's
  xmlreader API instead of the SAX2 API which has always felt a little
  hacky (see deleted comment block in `res_calendar_caldav.c`).

  The xmlreader API has been around since libxml2 2.5.0 which was
  released in 2003.

  Fixes #725


#### cdr_pgsql: Fix crash when the module fails to load multiple times.
  Author: chrsmj
  Date:   2024-05-16

  Missing or corrupt cdr_pgsql.conf configuration file can cause the
  second attempt to load the PostgreSQL CDR module to crash Asterisk via
  the Command Line Interface because a null CLI command is registered on
  the first failed attempt to load the module.

  Resolves: #736

#### asterisk.c: Don't log an error if .asterisk_history does not exist.
  Author: Sean Bright
  Date:   2024-05-27

  Fixes #751


#### chan_ooh323: Fix R/0 typo in docs
  Author: Walter Doekes
  Date:   2024-05-27


#### bundled_pjproject: Disable UPnP support.
  Author: Sean Bright
  Date:   2024-05-24

  Fixes #747


#### file.h: Rename function argument to avoid C++ keyword clash.
  Author: Sean Bright
  Date:   2024-05-24

  Fixes #744


#### rtp_engine: add support for multirate RFC2833 digits
  Author: Mike Bradeen
  Date:   2024-04-08

  Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.

  Asterisk currently treats RFC2833 Digits as a single rtp payload type
  with a fixed bitrate of 8K.  This change would expand that to 8, 16,
  24 and 32K.

  This requires checking the offered rtp types for any of these bitrates
  and then adding an offer for each (if configured for RFC2833.)  DTMF
  generation must also be changed in order to look at the current outbound
  codec in order to generate appropriately timed rtp.

  For cases where no outgoing audio has yet been sent prior to digit
  generation, Asterisk now has a concept of a 'preferred' codec based on
  offer order.

  On inbound calls Asterisk will mimic the payload types of the RFC2833
  digits.

  On outbound calls Asterisk will choose the next free payload types starting
  with 101.

  UserNote: No change in configuration is required in order to enable this
  feature. Endpoints configured to use RFC2833 will automatically have this
  enabled. If the endpoint does not support this, it should not include it in
  the SDP offer/response.

  Resolves: #699
@
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@


1.2
log
@Update to Asterisk 21.3.1:  various bug fixes and minor improvements

## Change Log for Release asterisk-21.3.1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9): res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests

### Commits By Author:

- ### George Joseph (1):
  - Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport ad..


## Change Log for Release asterisk-21.3.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)

### Summary:

- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip_logger: Preserve logging state on reloads.
  Issuing "pjsip reload" will no longer disable
  logging if it was previously enabled from the CLI.

- #### loader.c: Allow dependent modules to be unloaded recursively.
  In certain circumstances, modules with dependency relations
  can have their dependents automatically recursively unloaded and loaded
  again using the "module refresh" CLI command or the ModuleLoad AMI command.

- #### tcptls/iostream:  Add support for setting SNI on client TLS connections
  Secure websocket client connections now send SNI in
  the TLS client hello.

- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
  set identify_by=transport for the pjsip endpoint. Then
  use the existing 'match' option and the new 'transport' option of
  the identify.
  Fixes: #672

- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  this new feature let users match endpoints based on the
  indound SIP requests' URI. To do so, add 'request_uri' to the
  endpoint's 'identify_by' option. The 'match_request_uri' option of
  the identify can be an exact match for the entire request uri, or a
  regular expression (between slashes). It's quite similar to the
  header identifer.
  Fixes: #599

- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.

- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  When using the Originate AMI Action, we now can pass the PreDialGoSub
  parameter, instructing the asterisk to perform an subrouting at
  channel before call start. With this parameter an call initiated
  by AMI can request the channel to start the call automaticaly,
  adding a SIP header to using GoSUB, instructing to autoanswer
  the channel, and proceeding the outbuound extension executing.
  Exemple of an context to perform the previus indication:
  [addautoanswer]
  exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  exten => _s,n,Return()

- #### manager.c: Add CLI command to kick AMI sessions.
  The "manager kick session" CLI command now
  allows kicking a specified AMI session.

- #### chan_dahdi: Allow specifying waitfordialtone per call.
  "waitfordialtone" may now be specified for DAHDI
  trunk channels on a per-call basis using the CHANNEL function.

- #### Upgrade bundled pjproject to 2.14.1
  Bundled pjproject has been upgraded to 2.14.1. For more
  information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1


### Upgrade Notes:

- #### pbx_variables.c: Prevent SEGV due to stack overflow.
  The maximum amount of dialplan recursion
  using variable substitution (such as by using EVAL_EXTEN)
  is capped at 15.
@
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@


1.1
log
@comms/asterisk21: import asterisk-21.2.0

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 IAX.

This is an standard version.  It is secheduled to go to security
fixes only on November 18th, 2025, and EOL on November 18th, 2026.
See here for more information about Asterisk versions:
https://docs.asterisk.org/About-the-Project/Asterisk-Versions

Note that many things that have long been deprecated have now been
removed, such as chan_sip and app_macro.  See here for a complete
list:  http://docs.asterisk.org/Development/Asterisk-Module-Deprecations
@
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d3 12
a14 12
BLAKE2s (asterisk-21.2.0/asterisk-21.2.0.tar.gz) = f96a9cdf12c25b04b55ffa1250be62358b57e1b12e96dc3c37738b2adf23590b
SHA512 (asterisk-21.2.0/asterisk-21.2.0.tar.gz) = f6ede5da3a9f3fefa30aac2a55e3a06c9fb185f1a484cfc9b7e014ebcdf222e60cdb024b6b3f7daa1ce7a3f61c2a24c0135849ea669f086c6e86da08928536de
Size (asterisk-21.2.0/asterisk-21.2.0.tar.gz) = 26312154 bytes
BLAKE2s (asterisk-21.2.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f7e5fe212d7e7cdca14c52527a2552311ab7762c3f1464b09ddedc7c66aebde
SHA512 (asterisk-21.2.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 3f2f7bf3d5bce3544bc013f913c352f0204a3ce96239987403eb9dce8bc87e64a61d437762323a422a87b2fad1f3bf3e7a5f3d0d340f912a1b1dbfea9479d41d
Size (asterisk-21.2.0/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz) = 4253587 bytes
BLAKE2s (asterisk-21.2.0/pjproject-2.14.md5) = 8cbd12b39e930e684607202c7cff8036d1fa10c0070f8d5b50c1850095b8b28d
SHA512 (asterisk-21.2.0/pjproject-2.14.md5) = ea4c33f75a0cc4afcd7a1f7fd2a53423a182ef16a7abfe2189bce0ef0e6b956c6eb4f89aeabee714dfed0616bfd8f4aa98602b1faeda1671e00ff21a150283d2
Size (asterisk-21.2.0/pjproject-2.14.md5) = 166 bytes
BLAKE2s (asterisk-21.2.0/pjproject-2.14.tar.bz2) = 1e76da2fa893451c66f41e5c3dd2a7594494c0852947ef2146d6b642550604d0
SHA512 (asterisk-21.2.0/pjproject-2.14.tar.bz2) = 5b60a1033c9f09c0fbb5b132229fa99fdc103f8864a7cdeadf217ec47ff6bc784827826f595c4bc76816bc0fbf5cc6e9751cc4f6307eeecb7c7d8f5d0413d4ac
Size (asterisk-21.2.0/pjproject-2.14.tar.bz2) = 8376462 bytes
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