asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	aa23bd9bd7	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	3 years ago
Asterisk Development Team	211bfe1220	Update for 21.2.0-rc1	3 years ago
Naveen Albert	6e5a6c176d	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 `bdf9327a33`)	3 years ago
Naveen Albert	ae2fa8c5f0	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 `2ac9c8fb5c`)	3 years ago
Shaaah	d43d250e14	app_queue.c : fix "queue add member" usage string Fixing bracket placement in the "queue add member" cli usage string. (cherry picked from commit `6f99268f79`)	3 years ago
Naveen Albert	6af036a855	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 `190b6eafb3`)	3 years ago
Sean Bright	0aa4dbcae7	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 `1a94502551`)	3 years ago
George Joseph	0fe641503a	.github: Remove timeout-minutes from gatetests (cherry picked from commit `94176ecb88`)	3 years ago
George Joseph	c085753c2e	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 `1b94c90524`)	3 years ago
George Joseph	82df7454dc	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 `215424aa76`)	3 years ago
George Joseph	ddc6cc5465	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite (cherry picked from commit `d58ca20247`)	3 years ago
George Joseph	d7e262226f	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 `2e0d837e01`)	3 years ago
Sebastian Jennen	91df8b031c	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 `3c72bc8a7b`)	3 years ago
Shyju Kanaprath	c33cdfd49e	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 `07055dce94`)	3 years ago
Sean Bright	419feb11bd	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 `31ab82840b`)	3 years ago
George Joseph	1daec58db8	.github: Add force_cherry_pick option to Releaser (cherry picked from commit `2d1a4bab25`)	3 years ago
George Joseph	53931fac06	.github: Remove start_version from Releaser (cherry picked from commit `4077b4b340`)	3 years ago
romryz	e95611216f	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 (cherry picked from commit `a7a03bc294`)	3 years ago
Naveen Albert	31cb02dc01	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 (cherry picked from commit `6ddcdfce1f`)	3 years ago
George Joseph	3ca2bb5e84	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 (cherry picked from commit `adcfbcd50e`)	3 years ago
George Joseph	f72d97b7b1	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 (cherry picked from commit `a5ae546b88`)	3 years ago
Naveen Albert	52a80f974b	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 (cherry picked from commit `d0d09ef010`)	3 years ago
Ben Ford	f7d37df114	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 (cherry picked from commit `6efa51f512`)	3 years ago
Flole998	096243745c	res_pjsip_outbound_registration.c: Add User-Agent header override This introduces a setting for outbound registrations to override the global User-Agent header setting. Resolves: #515 UserNote: PJSIP outbound registrations now support a per-registration User-Agent header (cherry picked from commit `c7fc6ae362`)	3 years ago
cmaj	1377ac9e89	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. (cherry picked from commit `c863e0d77d`)	3 years ago
Joshua C. Colp	731fddd5d0	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 (cherry picked from commit `8ce69eda14`)	3 years ago
Mike Bradeen	b510d681a1	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. (cherry picked from commit `69fe814813`)	3 years ago
George Joseph	5c12fda94e	.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. (cherry picked from commit `9baf49497e`)	3 years ago
Naveen Albert	5e0f1bb5d2	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 (cherry picked from commit `ed39406838`)	3 years ago
Sean Bright	36184820bd	res_pjsip_t38.c: Permit IPv6 SDP connection addresses. The existing code prevented IPv6 addresses from being properly parsed. Fixes #558 (cherry picked from commit `841cd1480c`)	3 years ago
Brad Smith	14f84cc202	BuildSystem: Bump autotools versions on OpenBSD. Bump up to the more commonly used and modern versions of autoconf and automake. (cherry picked from commit `9deb4c679e`)	3 years ago
Brad Smith	30d05081d7	main/utils: Simplify the FreeBSD ast_get_tid() handling FreeBSD has had kernel threads for 20+ years. (cherry picked from commit `ec2c10689f`)	3 years ago
Sean Bright	2ec0b83e5b	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 (cherry picked from commit `9f20b4659f`)	3 years ago
Sean Bright	3b74538fcf	rtp_engine.c: Correct sample rate typo for L16/44100. Fixes #555 (cherry picked from commit `671b47cfda`)	3 years ago
Naveen Albert	dd90d4536f	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 (cherry picked from commit `874ee6e9aa`)	3 years ago
Naveen Albert	e65c8cede5	res_calendar_icalendar: Print iCalendar error on parsing failure. If libical fails to parse a calendar, print the error message it provdes. Resolves: #492 (cherry picked from commit `ef891529fa`)	3 years ago
Sean Bright	dbcd737302	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 (cherry picked from commit `03ad690276`)	3 years ago
Mike Bradeen	ab1a9fa7d1	app_voicemail_odbc: remove macrocontext from voicemail_messages table When app_macro was deprecated, the macrocontext column was removed from the INSERT statement but the binds were not renumbered. This broke the insert. This change removes the macrocontext column via alembic and re-numbers the existing columns in the INSERT. Fixes: #527 UserNote: The fix requires removing the macrocontext column from the voicemail_messages table in the voicemail database via alembic upgrade. UpgradeNote: The fix requires that the voicemail database be upgraded via alembic. Upgrading to the latest voicemail database via alembic will remove the macrocontext column from the voicemail_messages table. (cherry picked from commit `a22db8fd60`)	3 years ago
Naveen Albert	76d33df366	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 (cherry picked from commit `4b908f364d`)	3 years ago
PeterHolik	727a4cceec	chan_rtp.c: MulticastRTP missing refcount without codec option Fixes: #529 (cherry picked from commit `3b46872a8f`)	3 years ago
PeterHolik	8a999b6706	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak Fixes: asterisk#536 (cherry picked from commit `6fe045fd64`)	3 years ago
Naveen Albert	947ba375d7	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 (cherry picked from commit `67088b256d`)	3 years ago
Asterisk Development Team	2dccaa4830	Update for 21.1.0	3 years ago
Asterisk Development Team	2310df0ff6	Update for 21.1.0-rc2	3 years ago
Naveen Albert	e735ab8cfb	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	3 years ago
Asterisk Development Team	6b8dd72f50	Update for 21.1.0-rc1	3 years ago
George Joseph	077a1b171c	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 (cherry picked from commit `c31cd32b82`)	3 years ago
Naveen Albert	625826afd4	menuselect: Use more specific error message. Instead of using the same error message for missing dependencies and conflicts, be specific about what actually went wrong. Resolves: #520 (cherry picked from commit `7683259f37`)	3 years ago
Maximilian Fridrich	b7701ba973	res_pjsip_nat: Fix potential use of uninitialized transport details The ast_sip_request_transport_details must be zero initialized, otherwise this could lead to a SEGV. Resolves: #509 (cherry picked from commit `81188ada5f`)	3 years ago
Naveen Albert	c148203225	app_if: Fix faulty EndIf branching. This fixes faulty branching logic for the EndIf application. Instead of computing the next priority, which should be done for false conditionals or ExitIf, we should simply advance to the next priority. Resolves: #341 (cherry picked from commit `1bf4493371`)	3 years ago

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