6829 Commits (b23f089472600ccff34a7d6ce1ff6f328f51e52e)

Author	SHA1	Message	Date
George Joseph	f6b9d9e7d7	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.	2 years ago
Sean Bright	2b5673b68d	cli.c: `core show channels concise` is not really deprecated. ... Fixes #675	2 years ago
jonatascalebe	988986bdc9	manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action ... manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too. For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient. UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication: [addautoanswer] exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0) exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0) exten => _s,n,Return()	2 years ago
Naveen Albert	fc80bed5a7	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.	2 years ago
Naveen Albert	ef7788e0e4	manager.c: Add CLI command to kick AMI sessions. ... This adds a CLI command that can be used to manually kick specific AMI sessions. Resolves: #485 UserNote: The "manager kick session" CLI command now allows kicking a specified AMI session.	2 years ago
George Joseph	628f8d7a43	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.	2 years ago
Sebastian Jennen	ea8ead4e13	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	2 years ago
Naveen Albert	526a6e0ce4	dsp.c: Fix and improve potentially inaccurate log message. ... 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	2 years ago
George Joseph	29a273618d	pjsip show channelstats: Prevent possible segfault when faxing ... 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	2 years ago
George Joseph	6871d1cdfc	Reduce startup/shutdown verbose logging ... 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	2 years ago
Joshua C. Colp	edf54951be	utils: Make behavior of ast_strsep* match strsep. ... 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	2 years ago
Brad Smith	b0992fb771	main/utils: Simplify the FreeBSD ast_get_tid() handling ... FreeBSD has had kernel threads for 20+ years.	2 years ago
Sean Bright	f2ac526172	rtp_engine.c: Correct sample rate typo for L16/44100. ... Fixes #555	2 years ago
Naveen Albert	f1a9ec4703	manager.c: Fix erroneous reloads in UpdateConfig. ... 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	2 years ago
Naveen Albert	ac71e40042	logger: Fix linking regression. ... Commit 008731b0a4 caused a regression by resulting in logger.xml being compiled and linked into the asterisk binary in lieu of logger.c on certain platforms if Asterisk was compiled in dev mode. To fix this, we ensure the file has a unique name without the extension. Most existing .xml files have been named differently from any .c files in the same directory or did not pose this issue. channels/pjsip/dialplan_functions.xml does not pose this issue but is also being renamed to adhere to this policy. Resolves: #539	2 years ago
Naveen Albert	5cf75f9b2c	func_frame_trace: Add CLI command to dump frame queue. ... 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	2 years ago
George Joseph	09052bfa51	Revert "core & res_pjsip: Improve topology change handling." ... This reverts commit 315eb551db. Over the past year, we've had several reports of "topology storms" occurring where 2 external facing channels connected by one or more local channels and bridges will get themselves in a state where they continually send each other topology change requests. This usually manifests itself in no-audio calls and a flood of "Exceptionally long queue length" messages. It appears that this commit is the cause so we're reverting it for now until we can determine a more appropriate solution. Resolves: #530	2 years ago
Naveen Albert	fa3922a4d2	manager.c: Fix regression due to using wrong free function. ... Commit 424be34563 introduced a regression by calling ast_free on memory allocated by realpath. This causes Asterisk to abort when executing this function. Since the memory is allocated by glibc, it should be freed using ast_std_free. Resolves: #513	2 years ago
Ben Ford	424be34563	manager.c: Prevent path traversal with GetConfig. ... When using AMI GetConfig, it was possible to access files outside of the Asterisk configuration directory by using filenames with ".." and "./" even while live_dangerously was not enabled. This change resolves the full path and ensures we are still in the configuration directory before attempting to access the file.	2 years ago
Naveen Albert	183954bed3	config_options.c: Fix truncation of option descriptions. ... This increases the format width of option descriptions to avoid needless truncation for longer descriptions. Resolves: #428	2 years ago
Naveen Albert	ce1f4b3018	manager.c: Improve clarity of "manager show connected". ... Improve the "manager show connected" CLI command to clarify that the last two columns are permissions related, not counts, and use sufficient widths to consistently display these values. ASTERISK-30143 #close Resolves: #482	2 years ago
Naveen Albert	d1fb397cfc	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430	2 years ago
Sean Bright	008731b0a4	logger.c: Move LOG_GROUP documentation to dedicated XML file. ... The `get_documentation` awk script will only extract the first DOCUMENTATION block that it finds in a given file. This is by design (9bc2127) to prevent AMI event documentation from being pulled in to the core.xml documentation file. Because of this, the `LOG_GROUP` documentation added in 89709e2 was not being properly extracted and was missing fom the resulting XML documentation file. This commit moves the `LOG_GROUP` documentation to a separate `logger.xml` file.	2 years ago
Sean Bright	8c3ebf9747	config.c: Log #exec include failures. ... If the script referenced by `#exec` does not exist, writes anything to stderr, or exits abnormally or with a non-zero exit status, we log that to Asterisk's error logging channel. Additionally, write out a warning if the script produces no output. Fixes #259	2 years ago
Sean Bright	ca931c9436	app.c: Allow ampersands in playback lists to be escaped. ... Any function or application that accepts a `&`-separated list of filenames can now include a literal `&` in a filename by wrapping the entire filename in single quotes, e.g.: ``` exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world) ``` Fixes #172 UpgradeNote: Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`.	2 years ago
Sean Bright	31c44d0634	uri.c: Simplify ast_uri_make_host_with_port()	2 years ago
Sean Bright	611010d67d	res_http_websocket.c: Set hostname on client for certificate validation. ... Additionally add a `assert()` to in the TLS client setup code to ensure that hostname is set when it is supposed to be. Fixes #433	2 years ago
Matthew Fredrickson	21412fddcb	app_followme.c: Grab reference on nativeformats before using it ... Fixes a crash due to a lack of proper reference on the nativeformats object before passing it into ast_request(). Also found potentially similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c Fixes: #388	2 years ago
Naveen Albert	89709e2583	logger: Add channel-based filtering. ... This adds the ability to filter console logging by channel or groups of channels. This can be useful on busy systems where an administrator would like to analyze certain calls in detail. A dialplan function is also included for the purpose of assigning a channel to a group (e.g. by tenant, or some other metric). ASTERISK-30483 #close Resolves: #242 UserNote: The console log can now be filtered by channels or groups of channels, using the logger filter CLI commands.	2 years ago
George Joseph	f309ffad3d	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	2 years ago
Holger Hans Peter Freyther	69cf329681	stasis: Update the snapshot after setting the redirect ... The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest.	2 years ago
Holger Hans Peter Freyther	69590ba33e	ari: Provide the caller ID RDNIS for the channels ... Provide the caller ID RDNIS when available. This will allow an application to follow the redirect.	2 years ago
Brad Smith	5c22cfccd9	main/utils: Implement ast_get_tid() for OpenBSD ... Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid().	2 years ago
Naveen Albert	82086545cc	core_local: Fix local channel parsing with slashes. ... Currently, trying to call a Local channel with a slash in the extension will fail due to the parsing of characters after such a slash as being dial modifiers. Additionally, core_local is inconsistent and incomplete with its parsing of Local dial strings in that sometimes it uses the first slash and at other times it uses the last. For instance, something like DAHDI/5 or PJSIP/device is a perfectly usable extension in the dialplan, but Local channels in particular prevent these from being called. This creates inconsistent behavior for users, since using a slash in an extension is perfectly acceptable, and using a Goto to accomplish this works fine, but if specified through a Local channel, the parsing prevents this. This fixes this by explicitly parsing options from the last slash in the extension, rather than the first one, which doesn't cause an issue for extensions with slashes. ASTERISK-30013 #close Resolves: #248	2 years ago
Bastian Triller	8536ce8c38	func_json: Fix crashes for some types ... This commit fixes crashes in JSON_DECODE() for types null, true, false and real numbers. In addition it ensures that a path is not deeper than 32 levels. Also allow root object to be an array. Add unit tests for above cases.	2 years ago
Eduardo	91e368c485	codec_builtin: Use multiples of 20 for maximum_ms ... Some providers require a multiple of 20 for the maxptime or fail to complete calls, e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used. Resolves: #260	2 years ago
George Joseph	13ccbc1d08	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS ... Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing.	2 years ago
George Joseph	95313d4727	asterisk.c: Use the euid's home directory to read/write cli history ... The CLI .asterisk_history file is read from/written to the directory specified by the HOME environment variable. If the root user starts asterisk with the -U/-G options, or with runuser/rungroup set in asterisk.conf, the asterisk process is started as root but then it calls setuid/setgid to set the new user/group. This does NOT reset the HOME environment variable to the new user's home directory though so it's still left as "/root". In this case, the new user will almost certainly NOT have access to read from or write to the history file. * Added function process_histfile() which calls getpwuid(geteuid()) and uses pw->dir as the home directory instead of the HOME environment variable. * ast_el_read_default_histfile() and ast_el_write_default_histfile() have been modified to use the new process_histfile() function. Resolves: #337	2 years ago
Mike Bradeen	e06448353d	cel: add publish user event helper ... Add a wrapper function around ast_cel_publish_event that packs event and extras into a blob before publishing Resolves:#330	2 years ago
George Joseph	67fd66e2fe	file.c: Add ability to search custom dir for sounds ... To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory.	2 years ago
George Joseph	fde0e19658	make_buildopts_h, et. al. Allow adding all cflags to buildopts.h ... The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options.	2 years ago
Naveen Albert	e899a02465	pbx.c: Fix gcc 12 compiler warning. ... Resolves: #277	2 years ago
Maximilian Fridrich	fb234abd84	main/refer.c: Fix double free in refer_data_destructor + potential leak ... Resolves: #267	2 years ago
Joshua C. Colp	be3d8266da	manager: Tolerate stasis messages with no channel snapshot. ... In some cases I have yet to determine some stasis messages may be created without a channel snapshot. This change adds some tolerance to this scenario, preventing a crash from occurring.	2 years ago
Maximilian Fridrich	51a7b18038	core/ari/pjsip: Add refer mechanism ... This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.	2 years ago
Joshua C. Colp	806515597e	audiohook: Unlock channel in mute if no audiohooks present. ... In the case where mute was called on a channel that had no audiohooks the code was not unlocking the channel, resulting in a deadlock. Resolves: #233	2 years ago
Mike Bradeen	a55fd5e471	utils: add lock timestamps for DEBUG_THREADS ... Adds last locked and unlocked timestamps as well as a counter for the number of times the lock has been attempted (vs locked/unlocked) to debug output printed using the DEBUG_THREADS option. Resolves: #110	2 years ago
Jaco Kroon	73326fb4e0	tcptls: when disabling a server port, we should set the accept_fd to -1. ... If we don't set this to -1 if the structure can be potentially re-used later then it's possible that we'll issue a close() on an unrelated file descriptor, breaking asterisk in other interesting ways. I believe this to be an unlikely scenario, but it costs nothing to be safe. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
George Joseph	6b4f49c0df	build: Fix a few gcc 13 issues ... * gcc 13 is now catching when a function is declared as returning an enum but defined as returning an int or vice versa. Fixed a few in app.h, loader.c, stasis_message.c. * gcc 13 is also now (incorrectly) complaining of dangling pointers when assigning a pointer to a local char array to a char *. Had to change that to an ast_alloca. Resolves: #155	2 years ago
Ben Ford	cfde21c0c7	AMI: Add CoreShowChannelMap action. ... Adds a new AMI action (CoreShowChannelMap) that takes in a channel name and provides a list of all channels that are connected to that channel, following local channel connections as well. Resolves: #104 UserNote: New AMI action CoreShowChannelMap has been added.	2 years ago