asterisk

Commit Graph

Author	SHA1	Message	Date
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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago
Sean Bright	3b74538fcf	rtp_engine.c: Correct sample rate typo for L16/44100. Fixes #555 (cherry picked from commit `671b47cfda`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago
PeterHolik	727a4cceec	chan_rtp.c: MulticastRTP missing refcount without codec option Fixes: #529 (cherry picked from commit `3b46872a8f`)	1 year ago
PeterHolik	8a999b6706	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak Fixes: asterisk#536 (cherry picked from commit `6fe045fd64`)	1 year 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`)	1 year ago
Asterisk Development Team	2dccaa4830	Update for 21.1.0	1 year ago
Asterisk Development Team	2310df0ff6	Update for 21.1.0-rc2	1 year 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	1 year ago
Asterisk Development Team	6b8dd72f50	Update for 21.1.0-rc1	1 year 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`)	1 year 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`)	1 year 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`)	1 year 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`)	1 year ago
Naveen Albert	ba4a8de400	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 (cherry picked from commit `bb364fc61f`)	1 year ago
George Joseph	c0843b907a	doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories These should have been deleted after the release of 21.0.0 but were missed. (cherry picked from commit `df958a7d63`)	1 year ago
Naveen Albert	ce29be5536	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 (cherry picked from commit `d20c3e2f6f`)	1 year ago
Naveen Albert	6c33bf874d	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 (cherry picked from commit `09bd80c627`)	1 year ago
Sean Bright	d5fc671ae4	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. (cherry picked from commit `e001a1b6d3`)	1 year ago
Naveen Albert	f485d3cc8b	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 (cherry picked from commit `3bb34477d4`)	1 year ago
George Joseph	0fd8f9ca88	MergeApproved.yml: Remove unneeded concurrency The concurrency parameter on the MergeAndCherryPick job has been rmeoved. It was a hold-over from earlier days. (cherry picked from commit `751f8649fd`)	1 year ago
Maximilian Fridrich	b3cff31e1a	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. (cherry picked from commit `366dc1e99f`)	1 year ago
Sean Bright	0f5d624740	pbx_config.c: Don't crash when unloading module. `pbx_config` subscribes to manager events to capture the `FullyBooted` event but fails to unsubscribe if the module is loaded after that event fires. If the module is unloaded, a crash occurs the next time a manager event is raised. We now unsubscribe when the module is unloaded if we haven't already unsubscribed. Fixes #470 (cherry picked from commit `16a42b2aec`)	1 year ago
George Joseph	b10a8aa212	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 (cherry picked from commit `aec2453688`)	1 year ago
Sean Bright	b9a9e1e742	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. (cherry picked from commit `1d05e34d98`)	1 year ago
Matthew Fredrickson	dd79040125	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 (cherry picked from commit `e0bf65bde6`)	1 year ago
Sean Bright	fb289b0bad	res_pjsip_header_funcs.c: Check URI parameter length before copying. Fixes #477 (cherry picked from commit `002d6c2108`)	1 year ago
Sean Bright	1c617f9b01	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 (cherry picked from commit `b437cc3267`)	1 year ago
Sean Bright	8087a4ef2c	make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS. If LOCAL_MOD_SUBDIRS contains absolute paths, do not prefix them with the path to Asterisk's source tree. Fixes #142 (cherry picked from commit `5f0b568341`)	1 year ago
Sean Bright	77e8011291	app_voicemail.c: Completely resequence mailbox folders. Resequencing is a process that occurs when we open a voicemail folder and discover that there are gaps between messages (e.g. `msg0000.txt` is missing but `msg0001.txt` exists). Resequencing involves shifting the existing messages down so we end up with a sequential list of messages. Currently, this process stops after reaching a threshold based on the message limit (`maxmsg`) configured on the current folder. However, if `maxmsg` is lowered when a voicemail folder contains more than `maxmsg + 10` messages, resequencing will not run completely leaving the mailbox in an inconsistent state. We now resequence up to the maximum number of messages permitted by `app_voicemail` (currently hard-coded at 9999 messages). Fixes #86 (cherry picked from commit `fbe92dce2b`)	1 year ago
Naveen Albert	4e774da45a	sig_analog: Fix channel leak when mwimonitor is enabled. When mwimonitor=yes is enabled for an FXO port, the do_monitor thread will launch mwi_thread if it thinks there could be MWI on an FXO channel, due to the noise threshold being satisfied. This, in turns, calls analog_ss_thread_start in sig_analog. However, unlike all other instances where __analog_ss_thread is called in sig_analog, this call path does not properly set pvt->ss_astchan to the Asterisk channel, which means that the Asterisk channel is NULL when __analog_ss_thread starts executing. As a result, the thread exits and the channel is never properly cleaned up by calling ast_hangup. This caused issues with do_monitor on incoming calls, as it would think the channel was still owned even while receiving events, leading to an infinite barrage of warning messages; additionally, the channel would persist improperly. To fix this, the assignment is added to the call path where it is missing (which is only used for mwi_thread). A warning message is also added since previously there was no indication that __analog_ss_thread was exiting abnormally. This resolves both the channel leak and the condition that led to the warning messages. Resolves: #458 (cherry picked from commit `c930230a73`)	1 year ago
Sean Bright	72d631b7bd	res_rtp_asterisk.c: Update for OpenSSL 3+. In `5ac5c2b0` we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence deprecation warnings. This commit switches over to using non-deprecated API. (cherry picked from commit `05924e30f9`)	1 year ago
Sean Bright	9831c65f38	alembic: Update list of TLS methods available on ps_transports. Related to #221 and #222. Also adds `*.ini` to the `.gitignore` file in ast-db-manage for convenience. (cherry picked from commit `c7838a352a`)	1 year ago
Naveen Albert	bef9a9422d	func_channel: Expose previously unsettable options. Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442 (cherry picked from commit `9211fb5e97`)	1 year ago

1 2 3 4 5 ...

33827 Commits (f72d97b7b108a7db7dec9e0e38cdc854fd1a91dc) All Branches Search

33827 Commits (f72d97b7b108a7db7dec9e0e38cdc854fd1a91dc)

All Branches