Change Log for Release asterisk-21.2.0 ======================================== Links: ---------------------------------------- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: ---------------------------------------- - res_pjsip_stir_shaken.c: Add checks for missing parameters - app_dial: Add dial time for progress/ringing. - app_voicemail: Properly reinitialize config after unit tests. - app_queue.c : fix "queue add member" usage string - app_voicemail: Allow preventing mark messages as urgent. - res_pjsip: Use consistent type for boolean columns. - attestation_config.c: Use ast_free instead of ast_std_free - Makefile: Add stir_shaken/cache to directories created on install - Stir/Shaken Refactor - translate.c: implement new direct comp table mode - README.md: Removed outdated link - strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. - res_rtp_asterisk.c: Correct coefficient in MOS calculation. - dsp.c: Fix and improve potentially inaccurate log message. - pjsip show channelstats: Prevent possible segfault when faxing - Reduce startup/shutdown verbose logging - configure: Rerun bootstrap on modern platform. - Upgrade bundled pjproject to 2.14. - res_pjsip_outbound_registration.c: Add User-Agent header override - app_speech_utils.c: Allow partial speech results. - utils: Make behavior of ast_strsep* match strsep. - app_chanspy: Add 'D' option for dual-channel audio - app_if: Fix next priority calculation. - res_pjsip_t38.c: Permit IPv6 SDP connection addresses. - BuildSystem: Bump autotools versions on OpenBSD. - main/utils: Simplify the FreeBSD ast_get_tid() handling - res_pjsip_session.c: Correctly format SDP connection addresses. - rtp_engine.c: Correct sample rate typo for L16/44100. - manager.c: Fix erroneous reloads in UpdateConfig. - res_calendar_icalendar: Print iCalendar error on parsing failure. - app_confbridge: Don't emit warnings on valid configurations. - app_voicemail_odbc: remove macrocontext from voicemail_messages table - chan_dahdi: Allow MWI to be manually toggled on channels. - chan_rtp.c: MulticastRTP missing refcount without codec option - chan_rtp.c: Change MulticastRTP nameing to avoid memory leak - func_frame_trace: Add CLI command to dump frame queue. User Notes: ---------------------------------------- - ### app_dial: Add dial time for progress/ringing. The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received. - ### app_voicemail: Allow preventing mark messages as urgent. The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'. - ### Stir/Shaken Refactor Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. - ### Upgrade bundled pjproject to 2.14. Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases - ### res_pjsip_outbound_registration.c: Add User-Agent header override PJSIP outbound registrations now support a per-registration User-Agent header - ### app_speech_utils.c: Allow partial speech results. The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs. - ### app_chanspy: Add 'D' option for dual-channel audio The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored. - ### app_voicemail_odbc: remove macrocontext from voicemail_messages table The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade. - ### chan_dahdi: Allow MWI to be manually toggled on channels. The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440 Upgrade Notes: ---------------------------------------- - ### Stir/Shaken Refactor The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure. - ### app_voicemail_odbc: remove macrocontext from voicemail_messages table The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table. Closed Issues: ---------------------------------------- - #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates - #351: [improvement]: Refactor res_stir_shaken to use libjwt - #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14 - #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels - #492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure - #515: [improvement]: Implement option to override User-Agent-Header on a per-registration basis - #527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext. - #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls - #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue - #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no" - #560: [bug]: EndIf() causes next priority to be skipped - #565: [bug]: Application Read() returns immediately - #569: [improvement]: Add option to interleave input and output frames on spied channel - #572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not - #582: [improvement]: Reduce unneeded logging during startup and shutdown - #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions - #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received - #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active - #595: [improvement]: dsp.c: Fix and improve confusing warning message. - #597: [bug]: wrong MOS calculation - #601: [new-feature]: translate.c: implement new direct comp table mode (PR #585) - #619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent - #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault - #634: [bug]: make install doesn't create the stir_shaken cache directory - #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function - #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c Commits By Author: ---------------------------------------- - ### Ben Ford (1): - Upgrade bundled pjproject to 2.14. - ### Brad Smith (2): - main/utils: Simplify the FreeBSD ast_get_tid() handling - BuildSystem: Bump autotools versions on OpenBSD. - ### Flole998 (1): - res_pjsip_outbound_registration.c: Add User-Agent header override - ### George Joseph (6): - Reduce startup/shutdown verbose logging - pjsip show channelstats: Prevent possible segfault when faxing - Stir/Shaken Refactor - Makefile: Add stir_shaken/cache to directories created on install - attestation_config.c: Use ast_free instead of ast_std_free - res_pjsip_stir_shaken.c: Add checks for missing parameters - ### Joshua C. Colp (1): - utils: Make behavior of ast_strsep* match strsep. - ### Mike Bradeen (2): - app_voicemail_odbc: remove macrocontext from voicemail_messages table - app_chanspy: Add 'D' option for dual-channel audio - ### Naveen Albert (10): - func_frame_trace: Add CLI command to dump frame queue. - chan_dahdi: Allow MWI to be manually toggled on channels. - res_calendar_icalendar: Print iCalendar error on parsing failure. - manager.c: Fix erroneous reloads in UpdateConfig. - app_if: Fix next priority calculation. - configure: Rerun bootstrap on modern platform. - dsp.c: Fix and improve potentially inaccurate log message. - app_voicemail: Allow preventing mark messages as urgent. - app_voicemail: Properly reinitialize config after unit tests. - app_dial: Add dial time for progress/ringing. - ### PeterHolik (2): - chan_rtp.c: Change MulticastRTP nameing to avoid memory leak - chan_rtp.c: MulticastRTP missing refcount without codec option - ### Sean Bright (6): - app_confbridge: Don't emit warnings on valid configurations. - rtp_engine.c: Correct sample rate typo for L16/44100. - res_pjsip_session.c: Correctly format SDP connection addresses. - res_pjsip_t38.c: Permit IPv6 SDP connection addresses. - strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. - res_pjsip: Use consistent type for boolean columns. - ### Sebastian Jennen (1): - translate.c: implement new direct comp table mode - ### Shaaah (1): - app_queue.c : fix "queue add member" usage string - ### Shyju Kanaprath (1): - README.md: Removed outdated link - ### cmaj (1): - app_speech_utils.c: Allow partial speech results. - ### romryz (1): - res_rtp_asterisk.c: Correct coefficient in MOS calculation. Detail: ---------------------------------------- - ### res_pjsip_stir_shaken.c: Add checks for missing parameters Author: George Joseph Date: 2024-03-11 * Added checks for missing session, session->channel and rdata in stir_shaken_incoming_request. * Added checks for missing session, session->channel and tdata in stir_shaken_outgoing_request. Resolves: #645 - ### app_dial: Add dial time for progress/ringing. Author: Naveen Albert Date: 2024-02-08 Add a timeout option to control the amount of time to wait if no early media is received before giving up. This allows aborting early if the destination is not being responsive. Resolves: #588 UserNote: The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received. - ### app_voicemail: Properly reinitialize config after unit tests. Author: Naveen Albert Date: 2024-02-29 Most app_voicemail unit tests were not properly cleaning up after themselves after running. This led to test mailboxes lingering around in the system. It also meant that if any unit tests in app_voicemail that create mailboxes were executed and the module was not unloaded/loaded again prior to running the test_voicemail_vm_info unit test, Asterisk would segfault due to an attempt to copy a NULL string. The load_config test did actually have logic to reinitialize the config after the test. However, this did not work in practice since load_config() would not reload the config since voicemail.conf had not changed during the test; thus, additional logic has been added to ensure that voicemail.conf is truly reloaded, after any unit tests which modify the users list. This prevents the SEGV due to invalid mailboxes lingering around, and also ensures that the system state is restored to what it was prior to the tests running. Resolves: #629 - ### app_queue.c : fix "queue add member" usage string Author: Shaaah Date: 2024-01-23 Fixing bracket placement in the "queue add member" cli usage string. - ### app_voicemail: Allow preventing mark messages as urgent. Author: Naveen Albert Date: 2024-02-24 This adds an option to allow preventing callers from leaving messages marked as 'urgent'. Resolves: #619 UserNote: The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'. - ### res_pjsip: Use consistent type for boolean columns. Author: Sean Bright Date: 2024-02-27 This migrates the relevant schema objects from the `('yes', 'no')` definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')` one. Fixes #617 - ### attestation_config.c: Use ast_free instead of ast_std_free Author: George Joseph Date: 2024-03-05 In as_check_common_config, we were calling ast_std_free on raw_key but raw_key was allocated with ast_malloc so it should be freed with ast_free. Resolves: #636 - ### Makefile: Add stir_shaken/cache to directories created on install Author: George Joseph Date: 2024-03-04 The default location for the stir_shaken cache is /var/lib/asterisk/keys/stir_shaken/cache but we were only creating /var/lib/asterisk/keys/stir_shaken on istall. We now create the cache sub-directory. Resolves: #634 - ### Stir/Shaken Refactor Author: George Joseph Date: 2023-10-26 Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure. - ### translate.c: implement new direct comp table mode Author: Sebastian Jennen Date: 2024-02-25 The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio. This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing). - add new table mode - hide the 999999 comp values, as these only indicate an issue with transcoding - hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding) Resolves: #601 - ### README.md: Removed outdated link Author: Shyju Kanaprath Date: 2024-02-23 Removed outdated link http://www.quicknet.net from README.md cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21 - ### strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. Author: Sean Bright Date: 2024-02-17 If a dynamic string is created with an initial length of 0, `ast_str_buffer(…)` will return an invalid pointer. This was a secondary discovery when fixing #65. - ### res_rtp_asterisk.c: Correct coefficient in MOS calculation. Author: romryz Date: 2024-02-06 Media Experience Score relies on incorrect pseudo_mos variable calculation. According to forming an opinion section of the documentation, calculation relies on ITU-T G.107 standard: https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion ITU-T G.107 Annex B suggests to calculate MOS with a coefficient "seven times ten to the power of negative six", 7 * 10^(-6). which would mean 6 digits after the decimal point. Current implementation has 7 digits after the decimal point, which downrates the calls. Fixes: #597 - ### dsp.c: Fix and improve potentially inaccurate log message. Author: Naveen Albert Date: 2024-02-09 If ast_dsp_process is called with a codec besides slin, ulaw, or alaw, a warning is logged that in-band DTMF is not supported, but this message is not always appropriate or correct, because ast_dsp_process is much more generic than just DTMF detection. This logs a more generic message in those cases, and also improves codec-mismatch logging throughout dsp.c by ensuring incompatible codecs are printed out. Resolves: #595 - ### pjsip show channelstats: Prevent possible segfault when faxing Author: George Joseph Date: 2024-02-09 Under rare circumstances, it's possible for the original audio session in the active_media_state default_session to be corrupted instead of removed when switching to the t38/image media session during fax negotiation. This can cause a segfault when a "pjsip show channelstats" attempts to print that audio media session's rtp statistics. In these cases, the active_media_state topology is correctly showing only a single t38/image stream so we now check that there's an audio stream in the topology before attempting to use the audio media session to get the rtp statistics. Resolves: #592 - ### Reduce startup/shutdown verbose logging Author: George Joseph Date: 2024-01-31 When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582 - ### configure: Rerun bootstrap on modern platform. Author: Naveen Albert Date: 2024-02-12 The last time configure was run, it was run on a system that did not enable -std=gnu11 by default, which meant that the restrict qualifier would not be recognized on certain platforms. This regenerates the configure files from running bootstrap.sh, so that these should be recognized on all supported platforms. Resolves: #586 - ### Upgrade bundled pjproject to 2.14. Author: Ben Ford Date: 2024-02-05 Fixes: #406 UserNote: Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases - ### res_pjsip_outbound_registration.c: Add User-Agent header override Author: Flole998 Date: 2023-12-13 This introduces a setting for outbound registrations to override the global User-Agent header setting. Resolves: #515 UserNote: PJSIP outbound registrations now support a per-registration User-Agent header - ### app_speech_utils.c: Allow partial speech results. Author: cmaj Date: 2024-02-02 Adds 'p' option to SpeechBackground() application. With this option, when the app timeout is reached, whatever the backend speech engine collected will be returned as if it were the final, full result. (This works for engines that make partial results.) Resolves: #572 UserNote: The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs. - ### utils: Make behavior of ast_strsep* match strsep. Author: Joshua C. Colp Date: 2024-01-31 Given the scenario of passing an empty string to the ast_strsep functions the functions would return NULL instead of an empty string. This is counter to how strsep itself works. This change alters the behavior of the functions to match that of strsep. Fixes: #565 - ### app_chanspy: Add 'D' option for dual-channel audio Author: Mike Bradeen Date: 2024-01-31 Adds the 'D' option to app chanspy that causes the input and output frames of the spied channel to be interleaved in the spy output frame. This allows the input and output of the spied channel to be decoded separately by the receiver. If the 'o' option is also set, the 'D' option is ignored as the audio being spied is inherently one direction. Fixes: #569 UserNote: The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored. - ### app_if: Fix next priority calculation. Author: Naveen Albert Date: 2024-01-28 Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed a branching issue but "overshoots" when calculating the next priority. This fixes that; accompanying test suite tests have also been extended. Resolves: #560 - ### res_pjsip_t38.c: Permit IPv6 SDP connection addresses. Author: Sean Bright Date: 2024-01-29 The existing code prevented IPv6 addresses from being properly parsed. Fixes #558 - ### BuildSystem: Bump autotools versions on OpenBSD. Author: Brad Smith Date: 2024-01-27 Bump up to the more commonly used and modern versions of autoconf and automake. - ### main/utils: Simplify the FreeBSD ast_get_tid() handling Author: Brad Smith Date: 2024-01-27 FreeBSD has had kernel threads for 20+ years. - ### res_pjsip_session.c: Correctly format SDP connection addresses. Author: Sean Bright Date: 2024-01-27 Resolves a regression identified by @justinludwig involving the rendering of IPv6 addresses in outgoing SDP. Also updates `media_address` on PJSIP endpoints so that if we are able to parse the configured value as an IP we store it in a format that we can directly use later. Based on my reading of the code it appeared that one could configure `media_address` as: ``` [foo] type = endpoint ... media_address = [2001:db8::] ``` And that value would be blindly copied into the outgoing SDP without regard to its format. Fixes #541 - ### rtp_engine.c: Correct sample rate typo for L16/44100. Author: Sean Bright Date: 2024-01-28 Fixes #555 - ### manager.c: Fix erroneous reloads in UpdateConfig. Author: Naveen Albert Date: 2024-01-25 Currently, a reload will always occur if the Reload header is provided for the UpdateConfig action. However, we should not be doing a reload if the header value has a falsy value, per the documentation, so this makes the reload behavior consistent with the existing documentation. Resolves: #551 - ### res_calendar_icalendar: Print iCalendar error on parsing failure. Author: Naveen Albert Date: 2023-12-14 If libical fails to parse a calendar, print the error message it provdes. Resolves: #492 - ### app_confbridge: Don't emit warnings on valid configurations. Author: Sean Bright Date: 2024-01-21 The numeric bridge profile options `internal_sample_rate` and `maximum_sample_rate` are documented to accept the special values `auto` and `none`, respectively. While these values currently work, they also emit warnings when used which could be confusing for users. In passing, also ensure that we only accept the documented range of sample rate values between 8000 and 192000. Fixes #546 - ### app_voicemail_odbc: remove macrocontext from voicemail_messages table Author: Mike Bradeen Date: 2024-01-10 When app_macro was deprecated, the macrocontext column was removed from the INSERT statement but the binds were not renumbered. This broke the insert. This change removes the macrocontext column via alembic and re-numbers the existing columns in the INSERT. Fixes: #527 UserNote: The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade. UpgradeNote: The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table. - ### chan_dahdi: Allow MWI to be manually toggled on channels. Author: Naveen Albert Date: 2023-11-10 This adds a CLI command to manually toggle the MWI status of a channel, useful for troubleshooting or resetting MWI devices, similar to the capabilities offered with SIP messaging to manually control MWI status. UserNote: The 'dahdi set mwi' now allows MWI on channels to be manually toggled if needed for troubleshooting. Resolves: #440 - ### chan_rtp.c: MulticastRTP missing refcount without codec option Author: PeterHolik Date: 2024-01-15 Fixes: #529 - ### chan_rtp.c: Change MulticastRTP nameing to avoid memory leak Author: PeterHolik Date: 2024-01-16 Fixes: asterisk#536 - ### func_frame_trace: Add CLI command to dump frame queue. Author: Naveen Albert Date: 2024-01-12 This adds a simple CLI command that can be used for analyzing all frames currently queued to a channel. A couple log messages are also adjusted to be more useful in tracing bridging problems. Resolves: #533