33897 Commits (fad7a7525bb6f39b6130f19ff9196fd6f7ec7c9c)

 

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Author	SHA1	Message	Date
Shyju Kanaprath	5e1309f89a	README.md: Removed outdated link ... Removed outdated link http://www.quicknet.net from README.md cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21	2 years ago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.	2 years ago
George Joseph	e62939b183	.github: Add force_cherry_pick option to Releaser	2 years ago
George Joseph	d7de14600e	.github: Remove start_version from Releaser	2 years 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	2 years 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	2 years 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	2 years 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	2 years 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	2 years 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	2 years 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.	2 years 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	2 years 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.	2 years 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.	2 years 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	2 years 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	2 years ago
Brad Smith	d43d15b052	BuildSystem: Bump autotools versions on OpenBSD. ... Bump up to the more commonly used and modern versions of autoconf and automake.	2 years ago
Brad Smith	75dc596843	main/utils: Simplify the FreeBSD ast_get_tid() handling ... FreeBSD has had kernel threads for 20+ years.	2 years 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	2 years ago
Sean Bright	0defd65e63	rtp_engine.c: Correct sample rate typo for L16/44100. ... Fixes #555	2 years 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	2 years 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	2 years 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	2 years 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	2 years 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	2 years 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	2 years ago
PeterHolik	683da188c9	chan_rtp.c: MulticastRTP missing refcount without codec option ... Fixes: #529	2 years ago
PeterHolik	9044dcc80a	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak ... Fixes: asterisk#536	2 years 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	2 years ago
George Joseph	b71e338eea	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	494fb0b675	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	2 years ago
Maximilian Fridrich	85dd7cea8a	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	2 years ago
Naveen Albert	f4edb3b87e	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	2 years ago
Naveen Albert	d6245bcfa3	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
George Joseph	f00e026416	res_rtp_asterisk: Fix regression issues with DTLS client check ... * Since ICE candidates are used for the check and pjproject is required to use ICE, res_rtp_asterisk was failing to compile when pjproject wasn't available. The check is now wrapped with an #ifdef HAVE_PJPROJECT. * The rtp->ice_active_remote_candidates container was being used to check the address on incoming packets but that container doesn't contain peer reflexive candidates discovered during negotiation. This was causing the check to fail where it shouldn't. We now check against pjproject's real_ice->rcand array which will contain those candidates. * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where we weren't zeroing out sin->sin_zero before returning. This was causing ast_sockaddr_cmp() to always return false when one of the inputs was converted from a pj_sockaddr, even if both inputs had the same address and port. Resolves: #500 Resolves: #503 Resolves: #505	2 years ago
Gitea	dad0f5c89b	res_pjsip_header_funcs: Duplicate new header value, don't copy. ... When updating an existing header the 'update' code incorrectly just copied the new value into the existing buffer. If the new value exceeded the available buffer size memory outside of the buffer would be written into, potentially causing a crash. This change makes it so that the 'update' now duplicates the new header value instead of copying it into the existing buffer.	2 years ago
Mike Bradeen	d9d6e4b7cf	res_pjsip: disable raw bad packet logging ... Add patch to split the log level for invalid packets received on the signaling port. The warning regarding the packet will move to level 2 so that it can still be displayed, while the raw packet will be at level 4.	2 years ago
George Joseph	04658ddcde	res_rtp_asterisk.c: Check DTLS packets against ICE candidate list ... When ICE is in use, we can prevent a possible DOS attack by allowing DTLS protocol messages (client hello, etc) only from sources that are in the active remote candidates list. Resolves: GHSA-hxj9-xwr8-w8pq	2 years ago
Ben Ford	77876a5bf4	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	98186cba63	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	ba742bd9de	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
Sean Bright	7909aba37f	make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation. ... Although `make_xml_documentation`'s `print_dependencies` command was corrected by the previous fix (#461) for #142, the `create_xml` was not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.	2 years ago
Naveen Albert	8cc5fc2e39	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
George Joseph	74904a0b3a	MergeApproved.yml: Remove unneeded concurrency ... The concurrency parameter on the MergeAndCherryPick job has been rmeoved. It was a hold-over from earlier days.	2 years ago
Maximilian Fridrich	8d7665db7b	app_dial: Add option "j" to preserve initial stream topology of caller ... Resolves: #462 UserNote: The option "j" is now available for the Dial application which uses the initial stream topology of the caller to create the outgoing channels.	2 years ago
George Joseph	26c98bc68b	ast_coredumper: Increase reliability ... Instead of searching for the asterisk binary and the modules in the filesystem, we now get their locations, along with libdir, from the coredump itself... For the binary, we can use `gdb -c <coredump> ... "info proc exe"`. gdb can print this even without having the executable and symbols. Once we have the binary, we can get the location of the modules with `gdb ... "print ast_config_AST_MODULE_DIR` If there was no result then either it's not an asterisk coredump or there were no symbols loaded. Either way, it's not usable. For libdir, we now run "strings" on the note0 section of the coredump (which has the shared library -> memory address xref) and search for "libasteriskssl|libasteriskpj", then take the dirname. Since we're now getting everything from the coredump, it has to be correct as long as we're not crossing namespace boundaries like running asterisk in a docker container but trying to run ast_coredumper from the host using a shared file system (which you shouldn't be doing). There is still a case for using --asterisk-bin and/or --libdir: If you've updated asterisk since the coredump was taken, the binary, libraries and modules won't match the coredump which will render it useless. If you can restore or rebuild the original files that match the coredump and place them in a temporary directory, you can use --asterisk-bin, --libdir, and a new --moddir option to point to them and they'll be correctly captured in a tarball created with --tarball-coredumps. If you also use --tarball-config, you can use a new --etcdir option to point to what normally would be the /etc/asterisk directory. Also addressed many "shellcheck" findings. Resolves: #445	2 years ago
Sean Bright	2f405d232a	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
Matthew Fredrickson	058ead0ea5	res_odbc.c: Allow concurrent access to request odbc connections ... There are valid scenarios where res_odbc's connection pool might have some dead or stuck connections while others are healthy (imagine network elements/firewalls/routers silently timing out connections to a single DB and a single IP address, or a heterogeneous connection pool connected to potentially multiple IPs/instances of a replicated DB using a DNS front end for load balancing and one replica fails). In order to time out those unhealthy connections without blocking access to other parts of Asterisk that may attempt access to the connection pool, it would be beneficial to not lock/block access around the entire pool in _ast_odbc_request_obj2 while doing potentially blocking operations on connection pool objects such as the connection_dead() test, odbc_obj_connect(), or by dereferencing a struct odbc_obj for the last time and triggering a odbc_obj_disconnect(). This would facilitate much quicker and concurrent timeout of dead connections via the connection_dead() test, which could block potentially for a long period of time depending on odbc.ini or other odbc connector specific timeout settings. This also would make rapid failover (in the clustered DB scenario) much quicker. This patch changes the locking in _ast_odbc_request_obj2() to not lock around odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to lock around truly shared, non-immutable state like the connection_cnt member and the connections list on struct odbc_class. Fixes: #465	2 years ago
Sean Bright	0ed2e8200a	res_pjsip_header_funcs.c: Check URI parameter length before copying. ... Fixes #477	2 years ago
Sean Bright	461c3c60e2	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