asterisk

Commit Graph

Author	SHA1	Message	Date
Brad Smith	bf57478a26	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	b0992fb771	main/utils: Simplify the FreeBSD ast_get_tid() handling FreeBSD has had kernel threads for 20+ years.	1 year ago
Sean Bright	db945243e6	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	f2ac526172	rtp_engine.c: Correct sample rate typo for L16/44100. Fixes #555	1 year 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	1 year ago
Naveen Albert	f4845f756f	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	53fac14e41	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	0668e5494a	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.	1 year ago
Naveen Albert	a3be6a455f	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	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	1 year ago
PeterHolik	5216a1133d	chan_rtp.c: MulticastRTP missing refcount without codec option Fixes: #529	1 year ago
PeterHolik	25fec97428	chan_rtp.c: Change MulticastRTP nameing to avoid memory leak Fixes: asterisk#536	1 year 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	1 year 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	1 year ago
Naveen Albert	73997b39bd	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	1 year ago
Maximilian Fridrich	14bd1ceef6	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	1 year ago
Naveen Albert	58b16a538d	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	1 year 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	1 year ago
George Joseph	8c3ececb12	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	1 year ago
Gitea	a1ca026825	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.	1 year ago
Mike Bradeen	39760d109b	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.	1 year ago
George Joseph	d7d7764cb0	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	1 year 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.	1 year 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	1 year 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	1 year ago
Sean Bright	44a5b99cdd	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.	1 year 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	1 year ago
George Joseph	63364bfbf4	MergeApproved.yml: Remove unneeded concurrency The concurrency parameter on the MergeAndCherryPick job has been rmeoved. It was a hold-over from earlier days.	1 year ago
Maximilian Fridrich	3d7a7b1a47	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.	1 year ago
Sean Bright	1920639eab	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	1 year ago
George Joseph	1a0acabb8a	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	1 year 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.	1 year ago
Matthew Fredrickson	45da3ff9fa	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	1 year ago
Sean Bright	8d87d403bc	res_pjsip_header_funcs.c: Check URI parameter length before copying. Fixes #477	1 year 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	1 year ago
Sean Bright	4f52ed660d	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	1 year ago
Sean Bright	3026ac08ab	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	1 year ago
Naveen Albert	1ce9e1fa8f	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	1 year ago
Sean Bright	83636e4b92	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.	1 year ago
Sean Bright	07e378e2c0	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.	1 year ago
Naveen Albert	c4bf59b781	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	1 year 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`.	1 year ago
Sean Bright	31c44d0634	uri.c: Simplify ast_uri_make_host_with_port()	1 year ago
Sean Bright	0b744cef15	func_curl.c: Remove CURLOPT() plaintext documentation. I assume this was missed when initially converting to XML documentation and we've been kicking the can down the road since.	1 year 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	1 year ago
Sean Bright	91c8866ff3	live_ast: Add astcachedir to generated asterisk.conf. `astcachedir` (added in `b0842713`) was not added to `live_ast` so continued to point to the system `/var/cache` directory instead of the one in the live environment.	1 year ago
George Joseph	f28047db36	SECURITY.md: Update with correct documentation URL	1 year ago
Naveen Albert	5f115e425c	func_lock: Add missing see-also refs to documentation. Resolves: #423	1 year 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	1 year ago
Naveen Albert	44b955e237	configs: Improve documentation for bandwidth in iax.conf. This improves the documentation for the bandwidth setting in iax.conf by making it clearer what the ramifications of this setting are. It also changes the sample default from low to high, since only high is compatible with good codecs that people will want to use in the vast majority of cases, and this is a common gotcha that trips up new users. Resolves: #425	1 year ago

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