asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	d0b51f1ff7	res_parking: Fail gracefully if parking lot is full. Currently, if a parking lot is full, bridge setup returns -1, causing dialplan execution to terminate without TryExec. However, such failures should be handled more gracefully, the same way they are on other paths, as indicated by the module's author, here: http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html Now, callers will hear the parking failure announcement, and dialplan will continue, which is consistent with existing failure modes. Resolves: #624	1 year ago
Sean Bright	78471aa1a5	res_config_mysql.c: Support hostnames up to 255 bytes. Fixes #654	1 year ago
Sean Bright	2aa7f7a4dc	res_pjsip: Fix alembic downgrade for boolean columns. When downgrading, ensure that we don't touch columns that didn't actually change during upgrade.	1 year ago
Stanislav Abramenkov	929bc2693c	Upgrade bundled pjproject to 2.14.1 Fixes: asterisk#648 UserNote: Bundled pjproject has been upgraded to 2.14.1. For more information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1	1 year ago
Sean Bright	957cde97fd	alembic: Quote new MySQL keyword 'qualify.' Fixes #651	1 year ago
Maximilian Fridrich	f626af8e13	res_pjsip_session: Reset pending_media_state->read_callbacks In handle_negotiated_sdp the pending_media_state->read_callbacks must be reset before they are added in the SDP handlers in handle_negotiated_sdp_session_media. Otherwise, old callbacks for removed streams and file descriptors could be added to the channel and Asterisk would poll on non-existing file descriptors. Resolves: #611	1 year ago
Sean Bright	3c1a2daecd	res_monitor.c: Don't emit a warning about 'X' being unrecognized. Code was added in `030f7d41` to warn if an unrecognized option was passed to an application, but code in Monitor was taking advantage of the fact that the application would silently accept an invalid option. We now recognize the invalid option but we don't do anything if it's set. Fixes #639	1 year ago
George Joseph	61b5298cd3	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	1 year ago
George Joseph	f63c87c3f8	.github: Add PAT to PRSubmitActions/Add Reviewers	1 year ago
Naveen Albert	17d6a08beb	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.	1 year ago
Naveen Albert	564d22acc6	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	1 year ago
Shaaah	157dff87b3	app_queue.c : fix "queue add member" usage string Fixing bracket placement in the "queue add member" cli usage string.	1 year ago
Naveen Albert	166fc3c3d0	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'.	1 year ago
Sean Bright	39199aac56	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	1 year ago
George Joseph	a6070e39ec	.github: Remove timeout-minutes from gatetests	1 year ago
George Joseph	54464321a8	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	1 year ago
George Joseph	5a734afdde	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	1 year ago
George Joseph	224182b898	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite	1 year ago
George Joseph	181edcc3a3	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.	1 year ago
Sean Bright	6c3eaba7c6	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`	1 year ago
Sebastian Jennen	fe623aa081	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	1 year ago
Shyju Kanaprath	5e1309f89a	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	1 year ago
Sean Bright	164001b05c	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.	1 year ago
George Joseph	e62939b183	.github: Add force_cherry_pick option to Releaser	1 year ago
George Joseph	d7de14600e	.github: Remove start_version from Releaser	1 year ago
romryz	be96652612	res_rtp_asterisk.c: Correct coefficient in MOS calculation. 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	1 year ago
Naveen Albert	c6f6d1a3ac	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	1 year ago
George Joseph	1aaf9e6a1d	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	1 year ago
George Joseph	57d30c17b3	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	1 year ago
Naveen Albert	72f4cf6798	configure: Rerun bootstrap on modern platform. 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	1 year ago
Ben Ford	e6140eac0c	Upgrade bundled pjproject to 2.14. 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	1 year ago
cmaj	d2de6a2158	app_speech_utils.c: Allow partial speech results. 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.	1 year ago
Joshua C. Colp	5e09e2afeb	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	1 year ago
Mike Bradeen	f31b981d92	app_chanspy: Add 'D' option for dual-channel audio 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.	1 year ago
George Joseph	305111e954	.github: Update github-script to v7 and fix a rest bug Need to update the github-script to v7 to squash deprecation warnings. Also fixed the API name for github.rest.pulls.requestReviewers.	1 year ago
Naveen Albert	aa5b9d1d9f	app_if: Fix next priority calculation. Commit `fa3922a4d2` fixed a branching issue but "overshoots" when calculating the next priority. This fixes that; accompanying test suite tests have also been extended. Resolves: #560	1 year ago
Sean Bright	d32852ded9	res_pjsip_t38.c: Permit IPv6 SDP connection addresses. The existing code prevented IPv6 addresses from being properly parsed. Fixes #558	1 year ago
Brad Smith	d43d15b052	BuildSystem: Bump autotools versions on OpenBSD. Bump up to the more commonly used and modern versions of autoconf and automake.	1 year ago
Brad Smith	75dc596843	main/utils: Simplify the FreeBSD ast_get_tid() handling FreeBSD has had kernel threads for 20+ years.	1 year ago
Sean Bright	72523f19a9	res_pjsip_session.c: Correctly format SDP connection addresses. 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	1 year ago
Sean Bright	0defd65e63	rtp_engine.c: Correct sample rate typo for L16/44100. Fixes #555	1 year ago
Naveen Albert	57bb985b26	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	1 year ago
Naveen Albert	d1da985143	res_calendar_icalendar: Print iCalendar error on parsing failure. If libical fails to parse a calendar, print the error message it provdes. Resolves: #492	1 year ago
Sean Bright	9dd9615edf	app_confbridge: Don't emit warnings on valid configurations. 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	1 year ago
Mike Bradeen	d6c10075f8	app_voicemail: add NoOp alembic script to maintain sync Adding a NoOp alembic script for the voicemail database to maintain version sync with other branches. Fixes: #527	1 year ago
Naveen Albert	49ec57fc7c	chan_dahdi: Allow MWI to be manually toggled on channels. 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	1 year ago
Naveen Albert	395c674556	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	1 year ago
PeterHolik	683da188c9	chan_rtp.c: MulticastRTP missing refcount without codec option Fixes: #529	1 year ago
PeterHolik	9044dcc80a	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak Fixes: asterisk#536	1 year ago
Naveen Albert	6b82ed819c	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	1 year ago

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