asterisk

Commit Graph

Author	SHA1	Message	Date
Jeremy Lainé	85538b6d71	docs: Fix minor typo in MixMonitor AMI action The `Options` argument was erroneously documented as lowercase `options`.	4 months ago
Naveen Albert	5b2f36ffae	utils: Disable old style definition warnings for libdb. Newer versions of gcc now warn about old style definitions, such as those in libdb, which causes compilation failure with DEVMODE enabled. Ignore these warnings for libdb. Resolves: #1085	4 months ago
fabriziopicconi	4d9a73af71	rtp.conf.sample: Correct stunaddr example.	4 months ago
George Joseph	bbc0a609fc	docs: Add version information to ARI resources and methods. * Dump a git blame of each file in rest-api/api-docs. * Get the commit for each "resourcePath" and "httpMethod" entry. * Find the tags for each commit (same as other processes). * Insert a "since" array after each "resourcePath" and "httpMethod" entry.	4 months ago
George Joseph	77ef1789ad	res_pjsip_authenticator_digest: Make correct error messages appear again. When an incoming request can't be matched to an endpoint, the "artificial" auth object is used to create a challenge to return in a 401 response and we emit a "No matching endpoint found" log message. If the client then responds with an Authorization header but the request still can't be matched to an endpoint, the verification will fail and, as before, we'll create a challenge to return in a 401 response and we emit a "No matching endpoint found" log message. HOWEVER, because there WAS an Authorization header and it failed verification, we should have also been emitting a "Failed to authenticate" log message but weren't because there was a check that short-circuited that it if the artificial auth was used. Since many admins use the "Failed to authenticate" message with log parsers like fail2ban, those attempts were not being recognized as suspicious. Changes: * digest_check_auth() now always emits the "Failed to authenticate" log message if verification of an Authorization header failed even if the artificial auth was used. * The verification logic was refactored to be clearer about the handling of the return codes from verify(). * Comments were added clarify what return codes digest_check_auth() should return to the distributor and the implications of changing them. Resolves: #1095	4 months ago
George Joseph	6456eb7630	alembic: Database updates required. This commit doesn't actually change anything. It just adds the following upgrade notes that were omitted from the original commits. Resolves: #1097 UpgradeNote: 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.	4 months ago
Sean Bright	67e89b3e77	docs: Indent <since> tags. Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	4 months ago
George Joseph	8da4b05107	res_pjsip: Fix startup/reload memory leak in config_auth. An issue in config_auth.c:ast_sip_auth_digest_algorithms_vector_init() was causing double allocations for the two supported_algorithms vectors to the tune of 915 bytes. The leak only happens on startup and when a reload is done and doesn't get bigger with the number of auth objects defined. * Pre-initialized the two vectors in config_auth:auth_alloc(). * Removed the allocations in ast_sip_auth_digest_algorithms_vector_init(). * Added a note to the doc for ast_sip_auth_digest_algorithms_vector_init() noting that the vector passed in should be initialized and empty. * Simplified the create_artificial_auth() function in pjsip_distributor. * Set the vector initialization count to 0 in config_global:global_apply().	5 months ago
George Joseph	f6a193e87e	docs: Add version information to application and function XML elements * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	5 months ago
George Joseph	936e88512e	docs: Add version information to manager event instance XML elements * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized.	5 months ago
Joshua C. Colp	cf18bd6fd1	LICENSE: Update company name, email, and address.	5 months ago
Sean Bright	2d958812ad	res_prometheus.c: Set Content-Type header on /metrics response. This should resolve the Prometheus error: > Error scraping target: non-compliant scrape target sending blank Content-Type and no fallback_scrape_protocol specified for target. Resolves: #1075	5 months ago
George Joseph	d52cd705d3	README.md, asterisk.c: Update Copyright Dates	5 months ago
George Joseph	772221c82a	docs: Add version information to configObject and configOption XML elements Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation.	5 months ago
George Joseph	fb533fcc73	res_pjsip_authenticator_digest: Fix issue with missing auth and DONT_OPTIMIZE The return code fom digest_check_auth wasn't explicitly being initialized. The return code also wasn't explicitly set to CHALLENGE when challenges were sent. When optimization was turned off (DONT_OPTIMIZE), the compiler was setting it to "0"(CHALLENGE) which worked fine. However, with optimization turned on, it was setting it to "1" (SUCCESS) so if there was no incoming Authorization header, the function was returning SUCCESS to the distributor allowing the request to incorrectly succeed. The return code is now initialized correctly and is now explicitly set to CHALLENGE when we send challenges.	5 months ago
Naveen Albert	eca4ec6b5e	ast_tls_cert: Add option to skip passphrase for CA private key. Currently, the ast_tls_cert file is hardcoded to use the -des3 option for 3DES encryption, and the script needs to be manually modified to not require a passphrase. Add an option (-e) that disables encryption of the CA private key so no passphrase is required. Resolves: #1064	5 months ago
Naveen Albert	1b0cd8b10d	chan_iax2: Avoid unnecessarily backlogging non-voice frames. Currently, when receiving an unauthenticated call, we keep track of the negotiated format in the chosenformat, which allows us to later create the channel using the right format. However, this was not done for authenticated calls. This meant that in certain circumstances, if we had not yet received a voice frame from the peer, only certain other types of frames (e.g. text), there were no variables containing the appropriate frame. This led to problems in the jitterbuffer callback where we unnecessarily bailed out of retrieving a frame from the jitterbuffer. This was logic intentionally added in commit `73103bdcd5` in response to an earlier regression, and while this prevents crashes, it also backlogs legitimate frames unnecessarily. The abort logic was initially added because at this point in the code, we did not have the negotiated format available to us. However, it should always be available to us as a last resort in chosenformat, so we now pull it from there if needed. This allows us to process frames the jitterbuffer even if voicefmt and peerfmt aren't set and still avoid the crash. The failsafe logic is retained, but now it shouldn't be triggered anymore. Resolves: #1054	5 months ago
Allan Nathanson	7e58e8d4f6	config.c: fix #tryinclude being converted to #include on rewrite Correct an issue in ast_config_text_file_save2() when updating configuration files with "#tryinclude" statements. The API currently replaces "#tryinclude" with "#include". The API also creates empty template files if the referenced files do not exist. This change resolves these problems. Resolves: https://github.com/asterisk/asterisk/issues/920	5 months ago
Naveen Albert	abfa8206f3	sig_analog: Add Last Number Redial feature. This adds the Last Number Redial feature to simple switch. UserNote: Users can now redial the last number called if the lastnumredial setting is set to yes. Resolves: #437	5 months ago
George Joseph	15ed357747	docs: Various XML fixes * channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element. * main/bucket.c: Removed XML completely since the "bucket" and "file" objects are internal only with no config file. * main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf" and should have been "acl.conf" * res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element. * res/res_http_media_cache.c: Fixed the configFile element name. It was "http_media_cache.conf" and should have been "res_http_media_cache.conf".	5 months ago
Sean Bright	dafac1f6f4	strings.c: Improve numeric detection in `ast_strings_match()`. Essentially, we were treating 1234x1234 and 1234x5678 as 'equal' because we were able to convert the prefix of each of these strings to the same number. Resolves: #1028	5 months ago
George Joseph	3c867e6e6c	docs: Enable since/version handling for XML, CLI and ARI documentation * Added the "since" element to the XML configObject and configOption elements in appdocsxml.dtd. * Added the "Since" section to the following CLI output: ``` config show help <module> <object> config show help <module> <object> <option> core show application <app> core show function <func> manager show command <command> manager show event <event> agi show commands topic <topic> ``` * Refactored the commands above to output their sections in the same order: Synopsis, Since, Description, Syntax, Arguments, SeeAlso * Refactored the commands above so they all use the same pattern for writing the output to the CLI. * Fixed several memory leaks caused by failure to free temporary output buffers. * Added a "since" array to the mustache template for the top-level resources (Channel, Endpoint, etc.) and to the paths/methods underneath them. These will be added to the generated markdown if present. Example: ``` "resourcePath": "/api-docs/channels.{format}", "requiresModules": [ "res_stasis_answer", "res_stasis_playback", "res_stasis_recording", "res_stasis_snoop" ], "since": [ "18.0.0", "21.0.0" ], "apis": [ { "path": "/channels", "description": "Active channels", "operations": [ { "httpMethod": "GET", "since": [ "18.6.0", "21.8.0" ], "summary": "List all active channels in Asterisk.", "nickname": "list", "responseClass": "List[Channel]" }, ``` NOTE: No versioning information is actually added in this commit. Those will be added separately and instructions for adding and maintaining them will be published on the documentation site at a later date.	5 months ago
Artem Umerov	92d69897b7	logger.h: Fix build when AST_DEVMODE is not defined. Resolves: #1058	5 months ago
Sean Bright	032235bd29	dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's `media` argument. Resolves: #1023	5 months ago
Abdelkader Boudih	4b6d40bd66	samples: Use "asterisk" instead of "postgres" for username	5 months ago
Sean Bright	131682c2c5	manager: Add `<since>` tags for all AMI actions.	5 months ago
Steffen Arntz	d31b79b2f8	logger.c fix: malformed JSON template this typo was mentioned before, but never got fixed. https://community.asterisk.org/t/logger-cannot-log-long-json-lines-properly/87618/6	5 months ago
Sean Bright	e4635e817f	manager.c: Rename restrictedFile to is_restricted_file. Also correct the spelling of 'privileges.'	5 months ago
Abdelkader Boudih	7c40d82f02	res_config_pgsql: normalize database connection option with cel and cdr by supporting new options name	5 months ago
Stanislav Abramenkov	e95f97ad38	res_pjproject: Fix typo (OpenmSSL->OpenSSL) Fix typo (OpenmSSL->OpenSSL) mentioned by bkford in #972	5 months ago
George Joseph	7dc9d85f2b	Add SHA-256 and SHA-512-256 as authentication digest algorithms * Refactored pjproject code to support the new algorithms and added a patch file to third-party/pjproject/patches * Added new parameters to the pjsip auth object: * password_digest = <algorithm>:<digest> * supported_algorithms_uac = List of algorithms to support when acting as a UAC. * supported_algorithms_uas = List of algorithms to support when acting as a UAS. See the auth object in pjsip.conf.sample for detailed info. * Updated both res_pjsip_authenticator_digest.c (for UAS) and res_pjsip_outbound_authentocator_digest.c (UAC) to suport the new algorithms. The new algorithms are only available with the bundled version of pjproject, or an external version > 2.14.1. OpenSSL version 1.1.1 or greater is required to support SHA-512-256. Resolves: #948 UserNote: The SHA-256 and SHA-512-256 algorithms are now available for authentication as both a UAS and a UAC.	5 months ago
Allan Nathanson	1cb741df4a	config.c: retain leading whitespace before comments Configurations loaded with the ast_config_load2() API and later written out with ast_config_text_file_save2() will have any leading whitespace stripped away. The APIs should make reasonable efforts to maintain the content and formatting of the configuration files. This change retains any leading whitespace from comment lines that start with a ";". Resolves: https://github.com/asterisk/asterisk/issues/970	5 months ago
Ben Ford	e358ce0762	manager.c: Restrict ListCategories to the configuration directory. 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.	5 months ago
Sean Bright	6b3dadfbe2	config.c: Fix off-nominal reference leak. This was identified and fixed by @Allan-N in #918 but it is an important fix in its own right. The fix here is slightly different than Allan's in that we just move the initialization of the problematic AO2 container to where it is first used. Fixes #1046	5 months ago
Abdelkader Boudih	cc2b5ec88b	normalize contrib/ast-db-manage/queue_log.ini.sample	5 months ago
George Joseph	79427c1ac1	Add C++ Standard detection to configure and fix a new C++20 compile issue * The autoconf-archive package contains macros useful for detecting C++ standard and testing other C++ capabilities but that package was never included in the install_prereq script so many existing build environments won't have it. Even if it is installed, older versions won't newer C++ standards and will actually cause an error if you try to test for that version. To make it available for those environments, the ax_cxx_compile_stdcxx.m4 macro has copied from the latest release of autoconf-archive into the autoconf directory. * A convenience wrapper(ast_cxx_check_std) around ax_cxx_compile_stdcxx was also added so checking the standard version and setting the asterisk-specific PBX_ variables becomes a one-liner: `AST_CXX_CHECK_STD([std], [force_latest_std])`. Calling that with a version of `17` for instance, will set PBX_CXX17 to 0 or 1 depending on whether the current c++ compiler supports stdc++17. HAVE_CXX17 will also be 'defined" or not depending on the result. * C++ compilers hardly ever default to the latest standard they support. g++ version 14 for instance supports up to C++23 but only uses C++17 by default. If you want to use C++23, you have to add `-std=gnu++=23` to the g++ command line. If you set the second argument of AST_CXX_CHECK_STD to "yes", the macro will automatically keep the highest `-std=gnu++` value that worked and pass that to the Makefiles. * The autoconf-archive package was added to install_prereq for future use. * Updated configure.ac to use AST_CXX_CHECK_STD() to check for C++ versions 11, 14, 17, 20 and 23. * Updated configure.ac to accept the `--enable-latest-cxx-std` option which will set the second option to AST_CXX_CHECK_STD() to "yes". The default is "no". * ast_copy_string() in strings.h declares the 'sz' variable as volatile and does an `sz--` on it later. C++20 no longer allows the `++` and `--` increment and decrement operators to be used on variables declared as volatile however so that was changed to `sz -= 1`.	5 months ago
Naveen Albert	afd15274e7	chan_dahdi: Fix wrong channel state when RINGING recieved. Previously, when AST_CONTROL_RINGING was received by a DAHDI device, it would set its channel state to AST_STATE_RINGING. However, an analysis of the codebase and other channel drivers reveals RINGING corresponds to physical power ringing, whereas AST_STATE_RING should be used for audible ringback on the channel. This also ensures the correct device state is returned by the channel state to device state conversion. Since there seems to be confusion in various places regarding AST_STATE_RING vs. AST_STATE_RINGING, some documentation has been added or corrected to clarify the actual purposes of these two channel states, and the associated device state mapping. An edge case that prompted this fix, but isn't explicitly addressed here, is that of an incoming call to an FXO port. The channel state will be "Ring", which maps to a device state of "In Use", not "Ringing" as would be more intuitive. However, this is semantic, since technically, Asterisk is treating this the same as any other incoming call, and so "Ring" is the semantic state (put another way, Asterisk isn't ringing anything, like in the cases where channels are in the "Ringing" state). Since FXO ports don't currently support Call Waiting, a suitable workaround for the above would be to ignore the device state and instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state))) since it will be Ring if the FXO port is idle (but a call is ringing on it) and Up if the FXO port is actually in use. (In both cases, the device state would misleadingly be "In Use".) Resolves: #1029	5 months ago
Stanislav Abramenkov	2e8d8c4a08	Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016 UserNote: 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	5 months ago
George Joseph	5beb61317a	.github: Set exit 0 in CherryPick and Recheck workflow Cleanup steps	5 months ago
George Joseph	6fa7c40b57	gcc14: Fix issues caught by gcc 14 * test_message.c: Fix segfaults caused by passing NULL as an sprintf fmt.	5 months ago
George Joseph	e677ec8473	Header fixes for compiling C++ source files A few tweaks needed to be done to some existing header files to allow them to be compiled when included from C++ source files. logger.h had declarations for ast_register_verbose() and ast_unregister_verbose() which caused C++ issues but those functions were actually removed from logger.c many years ago so the declarations were just removed from logger.h.	5 months ago
George Joseph	2fa9dcd51c	Add ability to pass arguments to unit tests from the CLI Unit tests can now be passed custom arguments from the command line. For example, the following command would run the "mytest" test in the "/main/mycat" category with the option "myoption=54" `CLI> test execute category /main/mycat name mytest options myoption=54` You can also pass options to an entire category... `CLI> test execute category /main/mycat options myoption=54` Basically, everything after the "options" keyword is passed verbatim to the test which must decide what to do with it. * A new API ast_test_get_cli_args() was created to give the tests access to the cli_args->argc and cli_args->argv elements. * Although not needed for the option processing, a new macro ast_test_validate_cleanup_custom() was added to test.h that allows you to specify a custom error message instead of just "Condition failed". * The test_skel.c was updated to demonstrate parsing options and the use of the ast_test_validate_cleanup_custom() macro.	5 months ago
Kent	21dba60ff2	res_pjsip: Add new AOR option "qualify_2xx_only" Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify feature to mark a contact as available only if an OPTIONS request returns a 2XX response. If the option is not specified or is false, any response to the OPTIONS request marks the contact as available. UserNote: 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.	6 months ago
George Joseph	865c8b8232	.github: Change the run name for OnPRStateChangedPriv	6 months ago
Jaco Kroon	9b45103bc8	res_odbc: release threads from potential starvation. Whenever a slot is freed up due to a failed connection, wake up a waiter before failing. In the case of a dead connection there could be waiters, for example, let's say two threads tries to acquire objects at the same time, with one in the cached connections, one will acquire the dead connection, and the other will enter into the wait state. The thread with the dead connection will clear up the dead connection, and then attempt a re-acquire (at this point there cannot be cached connections else the other thread would have received that and tried to clean up), as such, at this point we're guaranteed that either there are no waiting threads, or that the maxconnections - connection_cnt threads will attempt to re-acquire connections, and then either succeed, using those connections, or failing, and then signalling to release more waiters. Also fix the pointer log for ODBC handle %p dead which would always reflect NULL. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 months ago
Sperl Viktor	d832aae1d7	app_queue: indicate the paused state of a dynamically added member in queue_log. Fixes: #1021	6 months ago
George Joseph	9a1180c6af	Allow C++ source files (as extension .cc) in the main directory Although C++ files (as extension .cc) have been handled in the module directories for many years, the main directory was missing one line in its Makefile that prevented C++ files from being recognised there.	6 months ago
Alexey Khabulyak	ced3b41327	format_gsm.c: Added mime type Sometimes it's impossible to get a file extension from URL (eg. http://example.com/gsm/your) so we have to rely on content-type header. Currenly, asterisk does not support content-type for gsm format(unlike wav). Added audio/gsm according to https://www.rfc-editor.org/rfc/rfc4856.html	6 months ago
Maksim Nesterov	c50dea93a5	func_uuid: Add a new dialplan function to generate UUIDs This function is useful for uniquely identifying calls, recordings, and other entities in distributed environments, as well as for generating an argument for the AudioSocket application.	6 months ago
Sperl Viktor	7b7df5d30e	app_queue: allow dynamically adding a queue member in paused state. Fixes: #1007 UserNote: use the p option of AddQueueMember() for paused member state. Optionally, use the r(reason) option to specify a custom reason for the pause.	6 months ago

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