asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	d8a0290d4a	Update for certified-20.7-cert1	2 years ago
George Joseph	4887f60e96	.github: Pass app_id and app_priv_key to AsteriskMergePR (cherry picked from commit `8b5dafe39c`)	2 years ago
George Joseph	de10a8fae1	.github: Change OnPRMergeApproved to use default token (cherry picked from commit `e28c7bc4b4`)	2 years ago
Sean Bright	0c92241e86	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. Fixes #785 (cherry picked from commit `2726265a4a`)	2 years ago
George Joseph	b3283912d2	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 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. (cherry picked from commit `59ed64627c`)	2 years ago
George Joseph	ad965dc835	security_agreement.c: Always add the Require and Proxy-Require headers 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 (cherry picked from commit `35528195b2`)	2 years ago
George Joseph	05d82da614	.github: Use ASTERISKTEAM_PAT for PR merging (cherry picked from commit `51edc5062d`)	2 years ago
George Joseph	208f055e5f	stasis_channels: Use uniqueid and name to delete old snapshots 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 (cherry picked from commit `bb4d470a2d`)	2 years ago
George Joseph	16ac05eb4d	.github: Replace PR workflows with stubs that call reusables The PR workflows now are just stubs that call reusable workflows located in the asterisk-ci-actions repo. (cherry picked from commit `e5a81346b0`)	2 years ago
George Joseph	0bafe17648	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions (cherry picked from commit `bafd37ed3e`)	2 years ago
George Joseph	277a06d450	.github: Add PAT to PRSubmitActions/Add Reviewers (cherry picked from commit `8ebd2bedd5`)	2 years ago
Sean Bright	bf61eab334	xml.c: Update deprecated libxml2 API usage. 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 (cherry picked from commit `b46f447530`)	2 years ago
George Joseph	cc5a7e1c19	.github: Add branches to workflow_dispatch for NightlyTests (cherry picked from commit `e65073d4df`)	2 years ago
Asterisk Development Team	b020a073bc	Update for certified-20.7-cert1-rc2	2 years ago
George Joseph	790f2860ef	stir_shaken: Fix memory leak, typo in config, tn canonicalization * Fixed possible memory leak in tn_config:tn_get_etn() where we weren't releasing etn if tn or eprofile were null. * We now canonicalize TNs before using them for lookups or adding them to Identity headers. * Fixed a typo in stir_shaken.conf.sample. Resolves: #716 (cherry picked from commit `200da52128`)	2 years ago
Spiridonov Dmitry	801b0ca253	sorcery.c: Fixed crash error when executing "module reload" Fixed crash error when cli "module reload". The error appears when compiling with res_prometheus and using the sorcery memory cache for registrations (cherry picked from commit `7210a03816`)	2 years ago
George Joseph	66d2ab529b	logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT If you're tracing a large function that may call another function multiple times in different circumstances, it can be difficult to see from the trace output exactly which location that function was called from. There's no good way to automatically determine the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT simply print out a trace line before and after the call. The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is that SCOPE_CALL ignores the function's return value (if any) where SCOPE_CALL_WITH_RESULT allows you to specify the type of the function's return value so it can be assigned to a variable. SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT and the "int" return type. (cherry picked from commit `951be188ba`)	2 years ago
George Joseph	57b93ba7a3	res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2) * OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use X509_get_pubkey. The difference is that X509_get_pubkey requires the caller to free the EVP_PKEY themselves so we now let RAII_VAR do that. * OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now wrap it in an ao2 object. * OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all the certs from an X509_STORE and there's no easy way to polyfill it so the CLI commands that list profiles will show a "not supported" message instead of listing the certs in a store. Resolves: #676 (cherry picked from commit `7bbc7b5a3a`)	2 years ago
Martin Nystroem	b496feae38	res_ari.c: Add additional output to ARI requests when debug is enabled When ARI debug is enabled the logs will now output http method and the uri. Fixes: #666 (cherry picked from commit `29106567e7`)	2 years ago
Sean Bright	a2276abdc5	alembic: Fix compatibility with SQLAlchemy 2.0+. SQLAlchemy 2.0 changed the way that commits/rollbacks are handled causing the final `UPDATE` to our `alembic_version_<whatever>` tables to be rolled back instead of committed. We now use one connection to determine which `alembic_version_<whatever>` table to use and another to run the actual migrations. This prevents the erroneous rollback. This change is compatible with both SQLAlchemy 1.4 and 2.0. (cherry picked from commit `42fbf5c22f`)	2 years ago
Naveen Albert	1cfe5e02a4	pbx_variables.c: Prevent SEGV due to stack overflow. It is possible for dialplan to result in an infinite recursion of variable substitution, which eventually leads to stack overflow. If we detect this, abort substitution and log an error for the user to fix the broken dialplan. Resolves: #480 UpgradeNote: The maximum amount of dialplan recursion using variable substitution (such as by using EVAL_EXTEN) is capped at 15. (cherry picked from commit `f9f36ca702`)	2 years ago
Sean Bright	4ecf3ea66a	res_pjsip: Fix alembic downgrade for boolean columns. When downgrading, ensure that we don't touch columns that didn't actually change during upgrade. (cherry picked from commit `e1f7778b4c`)	2 years ago
Sean Bright	ce4cdffe51	alembic: Quote new MySQL keyword 'qualify.' Fixes #651 (cherry picked from commit `1b45f481f0`)	2 years ago
Ivan Poddubny	f5107404c5	asterisk.c: Fix sending incorrect messages to systemd notify Send "RELOADING=1" instead of "RELOAD=1" to follow the format expected by systemd (see sd_notify(3) man page). Do not send STOPPING=1 in remote console mode: attempting to execute "asterisk -rx" by the main process leads to a warning if NotifyAccess=main (the default) or to a forced termination if NotifyAccess=all. (cherry picked from commit `d7e638f2c3`)	2 years ago
George Joseph	60c3f0273c	tcptls/iostream: Add support for setting SNI on client TLS connections If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello. (cherry picked from commit `5b9da1213c`)	2 years ago
George Joseph	fabed0f5e1	make_buildopts_h: Always include DETECT_DEADLOCKS Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must always be included in buildopts.h instead of only when ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise. Resolves: #719 (cherry picked from commit `7d69eafb5c`)	2 years ago
Sean Bright	6a629bee13	alembic: Correct NULLability of PJSIP id columns. Fixes #695 (cherry picked from commit `a3a0139b93`)	2 years ago
George Joseph	c10eb8a93a	rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup rtp_engine.c and stun.c were calling ast_register_cleanup which is skipped if any loadable module can't be cleanly unloaded when asterisk shuts down. Since this will always be the case, their cleanup functions never get run. In a practical sense this makes no difference since asterisk is shutting down but if you're in development mode and trying to use the leak sanitizer, the leaks from both of those modules clutter up the output. (cherry picked from commit `3767e14d15`)	2 years ago
George Joseph	4f2c1e4336	manager.c: Add missing parameters to Login documentation * Added the AuthType and Key parameters for MD5 authentication. * Added the Events parameter. Resolves: #689 (cherry picked from commit `d47641cbcf`)	2 years ago
George Joseph	0b6ad07bca	Fix incorrect application and function documentation references There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org. (cherry picked from commit `be6dee18dc`)	2 years ago
Asterisk Development Team	d9d47b5535	Update for certified-20.7-cert1-rc1	2 years ago
George Joseph	315d296b97	Initial commit for certified-20.7 (cherry picked from commit `f631eda4fd`)	2 years ago
Asterisk Development Team	af5acc405d	Update for 20.7.0	2 years ago
Asterisk Development Team	9b6c74e82a	Update for 20.7.0-rc2	2 years ago
George Joseph	d7597e5148	res_pjsip_stir_shaken.c: Add checks for missing parameters * 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	2 years ago
Asterisk Development Team	959babc472	Update for 20.7.0-rc1	2 years ago
Naveen Albert	40fd161575	app_dial: Add dial time for progress/ringing. 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. (cherry picked from commit `e4adc962ca`)	2 years ago
Naveen Albert	5ade71ec9a	app_voicemail: Properly reinitialize config after unit tests. 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 (cherry picked from commit `082966bdad`)	2 years ago
Shaaah	395bcab787	app_queue.c : fix "queue add member" usage string Fixing bracket placement in the "queue add member" cli usage string. (cherry picked from commit `c6ff7a6a51`)	2 years ago
Naveen Albert	3469e515de	app_voicemail: Allow preventing mark messages as urgent. 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'. (cherry picked from commit `3c42b141d7`)	2 years ago
Sean Bright	f8da5a2ba3	res_pjsip: Use consistent type for boolean columns. This migrates the relevant schema objects from the `('yes', 'no')` definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')` one. Fixes #617 (cherry picked from commit `93e40968c3`)	2 years ago
George Joseph	f02ac33364	.github: Remove timeout-minutes from gatetests (cherry picked from commit `9f182e9f58`)	2 years ago
George Joseph	44563b5043	attestation_config.c: Use ast_free instead of ast_std_free 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 (cherry picked from commit `267348bee3`)	2 years ago
George Joseph	efc2124594	Makefile: Add stir_shaken/cache to directories created on install 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 (cherry picked from commit `df0221b53d`)	2 years ago
George Joseph	be5fd6180c	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite (cherry picked from commit `eed8288d0c`)	2 years ago
George Joseph	fd27df9479	Stir/Shaken Refactor 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. (cherry picked from commit `e6c7f1aee0`)	2 years ago
Sean Bright	507c28f268	alembic: Synchronize alembic heads between supported branches. This adds a dummy migration to 18 and 20 so that our alembic heads are synchronized across all supported branches. In this case the migration we are stubbing (24c12d8e9014) is: `775352ee6c` (cherry picked from commit `d122225279`)	2 years ago
Sebastian Jennen	77808edbd4	translate.c: implement new direct comp table mode 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 (cherry picked from commit `a47acba99b`)	2 years ago
Shyju Kanaprath	2868366f00	README.md: Removed outdated link Removed outdated link http://www.quicknet.net from README.md cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21 (cherry picked from commit `667b5ee192`)	2 years ago
Sean Bright	02a17926c4	strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. 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. (cherry picked from commit `31fc2877cb`)	2 years ago

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33874 Commits (d8a0290d4a8b0ece3955f9c7b895659f8eba88f2) All Branches Search

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