--- /dev/null
+
+## Change Log for Release asterisk-21.4.0-rc1
+
+### Links:
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0-rc1.md)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0-rc1)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0-rc1.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
+
projects / thirdparty / asterisk.git / commitdiff
