Update for 22.0.0-rc1

.version

@@ -0,0 +1 @@
+22.0.0-rc1

CHANGES.md

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+ChangeLogs/ChangeLog-22.0.0-rc1.md

ChangeLogs/ChangeLog-22.0.0-rc1.md

@@ -0,0 +1,307 @@
+
+## Change Log for Release asterisk-22.0.0-rc1
+
+### Links:
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0-rc1.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0-rc1)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0-rc1.tar.gz)  
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  
+
+### Summary:
+
+- Commits: 19
+- Commit Authors: 8
+- Issues Resolved: 10
+- 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:
+
+- #### feat: ARI "ChannelToneDetected" event                                           
+  Setting the TONE_DETECT dialplan function on a channel
+  in ARI will now cause a ChannelToneDetected ARI event to be raised
+  when the specified tone is detected.
+
+
+### Upgrade Notes:
+
+- #### app_record: Add RECORD_TIME output variable.                                    
+  The RECORD_TIME variable now contains
+  the duration of Record() recordings in milliseconds.
+
+
+### Commit Authors:
+
+- Alexei Gradinari: (2)
+- George Joseph: (7)
+- Gibbz00: (1)
+- Joshua C. Colp: (2)
+- Mike Bradeen: (2)
+- Naveen Albert: (1)
+- Sean Bright: (2)
+- Tinet-Mucw: (2)
+
+## 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
+  - 548: [improvement]: Get Record() audio duration/length
+  - 763: [bug]: autoservice thread stuck in an endless sleep
+  - 811: [new-feature]: ARI channel tone detect events.
+  - 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
+
+### 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..
+
+- #### George Joseph (5):
+  - 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_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
+  - res_stir_shaken: Check for disabled before param validation
+
+- #### Mike Bradeen (1):
+  - 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 (2):
+  - app_chanspy.c: resolving the issue with audiohook direction read
+  - app_chanspy.c: resolving the issue writing frame to whisper audiohook.
+
+
+### Commit List:
+
+-  Prepare master for Asterisk 22
+-  Update issue guidelines link for bug reports.
+-  .github: NightlyAdmin now calls external CloseStaleIssuesAndPRs
+-  app_record: Add RECORD_TIME output variable.
+-  Revert "app_record: Add RECORD_TIME output variable."
+-  feat: ARI "ChannelToneDetected" event
+-  Update version for Asterisk 22
+-  res_stir_shaken: Check for disabled before param validation
+-  app_chanspy.c: resolving the issue writing frame to whisper audiohook.
+-  res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
+-  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.
+
+### Commit Details:
+
+#### Prepare master for Asterisk 22
+  Author: George Joseph
+  Date:   2023-08-09
+
+
+#### Update issue guidelines link for bug reports.
+  Author: Joshua C. Colp
+  Date:   2023-10-27
+
+
+#### .github: NightlyAdmin now calls external CloseStaleIssuesAndPRs
+  Author: George Joseph
+  Date:   2024-03-20
+
+
+#### app_record: Add RECORD_TIME output variable.
+  Author: Naveen Albert
+  Date:   2024-01-22
+
+  This adds the RECORD_TIME variable to Record(),
+  which is set to the recording duration before
+  the application returns.
+
+  Resolves: #548
+
+  UpgradeNote: The RECORD_TIME variable now contains
+  the duration of Record() recordings in milliseconds.
+
+#### Revert "app_record: Add RECORD_TIME output variable."
+  Author: Joshua C. Colp
+  Date:   2024-04-30
+
+  This reverts commit 6e8dccdbbf896bcc99046ae249db360698ede0b2.
+
+#### feat: ARI "ChannelToneDetected" event
+  Author: gibbz00
+  Date:   2024-07-18
+
+  A stasis event is now produced when using the TONE_DETECT dialplan
+  function. This event is published over ARI using the ChannelToneDetected
+  event. This change does not make it available over AMI.
+
+  Fixes: #811
+
+  UserNote: Setting the TONE_DETECT dialplan function on a channel
+  in ARI will now cause a ChannelToneDetected ARI event to be raised
+  when the specified tone is detected.
+
+#### Update version for Asterisk 22
+  Author: Mike Bradeen
+  Date:   2024-08-14
+
+
+#### 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
+
+#### 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
+
+#### 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
+
+

contrib/realtime/mysql/mysql_cdr.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start DATETIME, 
+    answer DATETIME, 
+    end DATETIME, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
+
+ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+

contrib/realtime/mysql/mysql_queue_log.sql

@@ -0,0 +1,29 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 4105ee839f58
+
+CREATE TABLE queue_log (
+    id BIGSERIAL NOT NULL, 
+    time TIMESTAMP WITHOUT TIME ZONE, 
+    callid VARCHAR(80), 
+    queuename VARCHAR(256), 
+    agent VARCHAR(80), 
+    event VARCHAR(32), 
+    data1 VARCHAR(100), 
+    data2 VARCHAR(100), 
+    data3 VARCHAR(100), 
+    data4 VARCHAR(100), 
+    data5 VARCHAR(100), 
+    PRIMARY KEY (id), 
+    UNIQUE (id)
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58') RETURNING alembic_version.version_num;
+
+COMMIT;
+

contrib/realtime/mysql/mysql_voicemail.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+-- Running upgrade 39428242f7f5 -> 1c55c341360f
+
+ALTER TABLE voicemail_messages DROP COLUMN macrocontext;
+
+UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
+

contrib/realtime/postgresql/postgresql_cdr.sql

@@ -0,0 +1,45 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start TIMESTAMP WITHOUT TIME ZONE, 
+    answer TIMESTAMP WITHOUT TIME ZONE, 
+    "end" TIMESTAMP WITHOUT TIME ZONE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d') RETURNING alembic_version.version_num;
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
+
+ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+
+COMMIT;
+

contrib/realtime/postgresql/postgresql_queue_log.sql

@@ -0,0 +1,29 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 4105ee839f58
+
+CREATE TABLE queue_log (
+    id BIGSERIAL NOT NULL, 
+    time TIMESTAMP WITHOUT TIME ZONE, 
+    callid VARCHAR(80), 
+    queuename VARCHAR(256), 
+    agent VARCHAR(80), 
+    event VARCHAR(32), 
+    data1 VARCHAR(100), 
+    data2 VARCHAR(100), 
+    data3 VARCHAR(100), 
+    data4 VARCHAR(100), 
+    data5 VARCHAR(100), 
+    PRIMARY KEY (id), 
+    UNIQUE (id)
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58') RETURNING alembic_version.version_num;
+
+COMMIT;
+

contrib/realtime/postgresql/postgresql_voicemail.sql

@@ -0,0 +1,45 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BYTEA, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e') RETURNING alembic_version.version_num;
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+-- Running upgrade 39428242f7f5 -> 1c55c341360f
+
+ALTER TABLE voicemail_messages DROP COLUMN macrocontext;
+
+UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
+
+COMMIT;
+