asterisk

Commit Graph

Author	SHA1	Message	Date
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.	2 years ago
George Joseph	f28047db36	SECURITY.md: Update with correct documentation URL	2 years ago
Naveen Albert	5f115e425c	func_lock: Add missing see-also refs to documentation. Resolves: #423	2 years 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	2 years 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	2 years ago
Naveen Albert	89709e2583	logger: Add channel-based filtering. This adds the ability to filter console logging by channel or groups of channels. This can be useful on busy systems where an administrator would like to analyze certain calls in detail. A dialplan function is also included for the purpose of assigning a channel to a group (e.g. by tenant, or some other metric). ASTERISK-30483 #close Resolves: #242 UserNote: The console log can now be filtered by channels or groups of channels, using the logger filter CLI commands.	2 years ago
Sean Bright	fb74ef1c7a	chan_iax2.c: Don't send unsanitized data to the logger. This resolves an issue where non-printable characters could be sent to the console/log files.	2 years ago
George Joseph	36f8490901	codec_ilbc: Disable system ilbc if version >= 3.0.0 Fedora 37 started shipping ilbc 3.0.4 which we don't yet support. configure.ac now checks the system for "libilbc < 3" instead of just "libilbc". If true, the system version of ilbc will be used. If not, the version included at codecs/ilbc will be used. Resolves: #84	2 years ago
Sean Bright	c19470497f	resource_channels.c: Explicit codec request when creating UnicastRTP. Fixes #394	2 years ago
Sean Bright	c040179fcf	doc: Update IP Quality of Service links. Fixes #328	2 years ago
George Joseph	f309ffad3d	chan_pjsip: Add PJSIPHangup dialplan app and manager action See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	2 years ago
Sean Bright	52968038ed	chan_iax2.c: Ensure all IEs are displayed when dumping frame contents. When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE in a frame was one that may not have any data - such as the CALLTOKEN IE in an NEW request - it was not getting displayed.	2 years ago
Naveen Albert	293577f1d9	chan_dahdi: Warn if nonexistent cadence is requested. If attempting to ring a channel using a nonexistent cadence, emit a warning, before falling back to the default cadence. Resolves: #409	2 years ago
Holger Hans Peter Freyther	69cf329681	stasis: Update the snapshot after setting the redirect The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest.	2 years ago
Holger Hans Peter Freyther	69590ba33e	ari: Provide the caller ID RDNIS for the channels Provide the caller ID RDNIS when available. This will allow an application to follow the redirect.	2 years ago
Brad Smith	5c22cfccd9	main/utils: Implement ast_get_tid() for OpenBSD Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid().	2 years ago
Brad Smith	6ec59e1e04	res_rtp_asterisk.c: Fix runtime issue with LibreSSL The module will fail to load. Use proper function DTLS_method() with LibreSSL.	2 years ago
Naveen Albert	4657163c56	app_directory: Add ADSI support to Directory. This adds optional ADSI support to the Directory application, which allows callers with ADSI CPE to navigate the Directory system significantly faster than is possible using the audio prompts. Callers can see the directory name (and optionally extension) on their screenphone and confirm or reject a match immediately rather than waiting for it to be spelled out, enhancing usability. Resolves: #356	2 years ago
Naveen Albert	82086545cc	core_local: Fix local channel parsing with slashes. Currently, trying to call a Local channel with a slash in the extension will fail due to the parsing of characters after such a slash as being dial modifiers. Additionally, core_local is inconsistent and incomplete with its parsing of Local dial strings in that sometimes it uses the first slash and at other times it uses the last. For instance, something like DAHDI/5 or PJSIP/device is a perfectly usable extension in the dialplan, but Local channels in particular prevent these from being called. This creates inconsistent behavior for users, since using a slash in an extension is perfectly acceptable, and using a Goto to accomplish this works fine, but if specified through a Local channel, the parsing prevents this. This fixes this by explicitly parsing options from the last slash in the extension, rather than the first one, which doesn't cause an issue for extensions with slashes. ASTERISK-30013 #close Resolves: #248	2 years ago
Mark Murawski	fcb9feddfd	Remove files that are no longer updated Fixes: #360	2 years ago
Naveen Albert	d678370b54	app_voicemail: Add AMI event for mailbox PIN changes. This adds an AMI event that is emitted whenever a mailbox password is successfully changed, allowing AMI consumers to process these. UserNote: The VoicemailPasswordChange event is now emitted whenever a mailbox password is updated, containing the mailbox information and the new password. Resolves: #398	2 years ago
Sean Bright	6c6137028b	app_queue.c: Emit unpause reason with PauseQueueMember event. Fixes #395	2 years ago
George Joseph	ed4978ae8b	bridge_simple: Suppress unchanged topology change requests In simple_bridge_join, we were sending topology change requests even when the new and old topologies were the same. In some circumstances, this can cause unnecessary re-invites and even a re-invite flood. We now suppress those. Resolves: #384	2 years ago
Naveen Albert	b94f8bb216	res_pjsip: Include cipher limit in config error message. If too many ciphers are specified in the PJSIP config, include the maximum number of ciphers that may be specified in the user-facing error message. Resolves: #396	2 years ago
Mike Bradeen	f666dd0dd8	res_speech: allow speech to translate input channel * Allow res_speech to translate the input channel if the format is translatable to a format suppored by the speech provider. Resolves: #129 UserNote: res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites.	2 years ago

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