asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	4efea4780b	manager: Fix incomplete filtering of AMI events. The global event filtering code was only in one possible execution path, so not all events were being properly filtered out if requested. This moves that into the universal AMI handling code so all events are properly handled. Additionally, the CLI listing of disabled events can also get truncated, so we now print out everything. ASTERISK-30137 #close Change-Id: If8c42edcb2abc5158552da7eba2a8ff6b20e1959	3 years ago
Naveen Albert	e8205fb166	db: Add AMI action to retrieve DB keys at prefix. Adds the DBGetTree action, which can be used to retrieve all of the DB keys beginning with a particular prefix, similar to the capability provided by the database show CLI command. ASTERISK-30136 #close Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6	3 years ago
Michael Neuhauser	6fbcb24abb	res_pjsip: delay contact pruning on Asterisk start Move the call to ast_sip_location_prune_boot_contacts() after the call to ast_res_pjsip_init_options_handling() so that res/res_pjsip/pjsip_options.c is informed about the contact deletion and updates its sip_options_contact_statuses list. This allows for an AMI event to be sent by res/res_pjsip/pjsip_options.c if the endpoint registers again from the same remote address and port (i.e., same URI) as used before the Asterisk restart. ASTERISK-30109 Reported-by: Michael Neuhauser Change-Id: I1ba4478019e4931a7085f62708d9b66837e901a8	3 years ago
Sam Banks	30cc5bab2f	queues.conf.sample: Correction of typo ASTERISK-30126 #close Change-Id: I009c4dcbf9338a13e3baf87b52a5bbe4f9f81a42	3 years ago
Naveen Albert	5177bb9f34	chan_dahdi: Fix buggy and missing Caller ID parameters There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in 67 and 82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch 67/82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15	3 years ago
Naveen Albert	15dc2f8c16	chan_dahdi: Add POLARITY function. Adds a POLARITY function which can be used to retrieve the current polarity of an FXS channel as well as set the polarity of an FXS channel to idle or reverse at any point during a call. ASTERISK-30000 #close Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1	3 years ago
Naveen Albert	ebe9b566d2	app_confbridge: Always set minimum video update interval. Currently, if multiple video-enabled ConfBridges are conferenced together, we immediately get into a scenario where an infinite sequence of video updates fills up the taskprocessor queue and causes memory consumption to climb unabated until Asterisk is killed. This is due to the core bridging mechanism that provides video updates (softmix_bridge_write_control in bridge_softmix.c) continously updating all the channels in the bridge with video updates. The logic to do so in the core is that the video updates should be provided if the video_update_discard property for the bridge is 0, or if enough time has elapsed since the last video update. Thus, we already have a safeguard built in to ensure the scenario described above does not happen. Currently, however, this safeguard is not being adequately ensured. In app_confbridge, the video_update_discard property defaults to 2000, which is a healthy value that should completely prevent this issue. However, this value is only set onto the bridge in the SFU video mode. This leaves video modes such as follow_talker completely vulnerable, since video_update_discard will actually be 0, since the default or set value was never applied. As a result, the core bridging mechanism will always try to provide video updates regardless of when the last one was sent. To prevent this issue from happening, we now always set the video_update_discard property on the bridge with the value from the bridge profile. The app_confbridge defaults will thus ensure that infinite video updates no longer happen in any video mode. ASTERISK-29907 #close Change-Id: I4accb2536ac62797950468e9930f12ef7dd486b2	3 years ago
Mike Bradeen	306186092e	Makefile: Avoid git-make user conflict make_version now silently checks if the required git commands will fail. If they do, then return UNKNOWN__git_check_fail to distinguish this failure from other UNKNOWN__ version failures Makefile checks for this value on install and exits out with instructions ASTERISK-30029 Change-Id: If8f10cac8f509c08981120f17555762342020221	3 years ago
Sean Bright	a0c3eb3b08	pbx.c: Simplify ast_context memory management. Allocate all of the ast_context's character data in the structure's flexible array member and eliminate the clunky fake_context. This will simplify future changes to ast_context. Change-Id: I98357de75d8ac2b3c4c9f201223632e6901021ea	3 years ago
George Joseph	dd0badf16b	geoloc_eprofile.c: Fix setting of loc_src in set_loc_src() line 196: loc_src = '\0'; should have been line 196: *loc_src = '\0'; The issue was caught by the gcc optimizer complaining that loc_src had a zero length because the pointer itself was being set to NULL instead of the _contents_ of the pointer being set to the NULL terminator. ASTERISK-30138 Reported-by: Sean Bright Change-Id: Id247be113cc8510f043ca053d5b4f5f3d32acd29	3 years ago
George Joseph	a6f0fb9bf7	Geolocation: chan_pjsip Capability Preview This commit adds res_pjsip_geolocation which gives chan_pjsip the ability to use the core geolocation capabilities. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30128 Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9	3 years ago
George Joseph	916dd96c86	Geolocation: Core Capability Preview This commit adds res_geolocation which creates the core capabilities to manipulate Geolocation information on SIP INVITEs. An upcoming commit will add res_pjsip_geolocation which will allow the capabilities to be used with the pjsip channel driver. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30127 Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303	3 years ago
Naveen Albert	a8395c9420	general: Fix various typos. ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Kevin Harwell	9ba2f083cf	cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type See also: ASTERISK_30023 ASTERISK-30096 #close patches: inline on issue - submitted by Morvai Szabolcs Change-Id: I79c0b74862100acd9c8319dca5cc456a654d02eb	3 years ago
Naveen Albert	f812dfb68c	chan_iax2: Allow compiling without OpenSSL. ASTERISK_30007 accidentally made OpenSSL a required depdendency. This adds an ifdef so the relevant code is compiled only if OpenSSL is available, since it only needs to be executed if OpenSSL is available anyways. ASTERISK-30083 #close Change-Id: Iad05c1a9a8bd2a48e7edf8d234eaa9f80779e34d	3 years ago
Naveen Albert	8a0ff5c0a6	res_cliexec: Add dialplan exec CLI command. Adds a CLI command similar to "dialplan eval function" except for applications: "dialplan exec application", useful for quickly testing certain application behavior directly from the CLI without writing any dialplan. ASTERISK-30062 #close Change-Id: I42e9fa9b60746c21450d40f99a026d48d2486dde	3 years ago
Trevor Peirce	14144af7a6	features: Update documentation for automon and automixmon The current documentation is out of date and does not reflect actual behaviour. This change makes documentation clearer and accurately reflect the purpose of relevant channel variables. ASTERISK-30123 Change-Id: I160d0b01fce862477ad55ac1aa708a730473eb6f	3 years ago
George Joseph	d0ef2164d9	Geolocation: Base Asterisk Prereqs * Added ast_variable_list_from_quoted_string() Parse a quoted string into an ast_variable list. * Added ast_str_substitute_variables_full2() Perform variable/function/expression substitution on an ast_str. * Added ast_strsep_quoted() Like ast_strsep except you can specify a specific quote character. Also added unit test. * Added ast_xml_find_child_element() Find a direct child element by name. * Added ast_xml_doc_dump_memory() Dump the specified document to a buffer * ast_datastore_free() now checks for a NULL datastore before attempting to destroy it. Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec	3 years ago
Boris P. Korzun	5761d7ab3b	pbx_lua: Remove compiler warnings Improved variable definitions (specified correct type) for avoiding compiler warnings. ASTERISK-30117 #close Change-Id: I3b00c1befb658ee9379ddabd9a9132765ca9201a	3 years ago
Jose Lopes	043627f814	res_pjsip_header_funcs: Add functions PJSIP_RESPONSE_HEADER and PJSIP_RESPONSE_HEADERS These new functions allow retrieving information from headers on 200 OK INVITE response. ASTERISK-29999 Change-Id: I264a610a9333359297a0825feb29a1bb4f4ad144	3 years ago
Naveen Albert	51a79fd4fd	db: Notify user if deleted DB entry didn't exist. Currently, if using the CLI to delete a DB entry, "Database entry removed" is always returned, regardless of whether or not the entry actually existed in the first place. This meant that users were never told if entries did not exist. The same issue occurs if trying to delete a DB key using AMI. To address this, new API is added that is more stringent in deleting values from AstDB, which will not return success if the value did not exist in the first place, and will print out specific error details if available. ASTERISK-30001 #close Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18	3 years ago
Naveen Albert	12f693d401	cli: Fix CLI blocking forever on terminating backslash A corner case exists in CLI parsing where if a CLI user in a remote console ends with a backslash and then invokes command completion (using TAB or ?), then the console will freeze forever until a SIGQUIT signal is sent to the process, due to getting blocked forever reading the command completion. CTRL+C and other key combinations have no impact on the CLI session. This occurs because, in such cases, the CLI process is waiting for AST_CLI_COMPLETE_EOF to appear in the buffer from the main process, but instead the main process is confused by the funny syntax and thus prints out the CLI help. As a result, the CLI process is stuck on the read call, waiting for the completion that will never come. This prevents blocking forever by checking if the data from the main process starts with "Usage:". If it does, that means that CLI help was sent instead of the tab complete vector, and thus the CLI should bail out and not wait any longer. ASTERISK-29822 #close Change-Id: I9810ac59304fec162da701653c9c834f0ec8f670	3 years ago
Naveen Albert	2a8d407935	res_calendar_icalendar: Send user agent in request. Microsoft recently began rejecting all requests for ICS calendars on Office 365 with 400 errors if the request doesn't contain a user agent. See: https://docs.microsoft.com/en-us/answers/questions/883904/34the-remote-server-returned-an-error-400-bad-requ.html Accordingly, we now send a user agent on requests for ICS files so that requests to Office 365 will work as they did before. ASTERISK-30106 Change-Id: Ie9dcaef12ae8adf37533c684499eb11005fac8f7	3 years ago
Naveen Albert	f6f4e11d17	app_dial: Propagate outbound hook flashes. The Dial application currently stops hook flashes dead in their tracks from propagating through on outbound calls. This fixes that so they can go down the wire. ASTERISK-30115 #close Change-Id: Id4e78b29a049f35c5b1e7520eaa10d0eb5b7f97c	3 years ago
Naveen Albert	96e56b89e0	app_dial: Fix dial status regression. ASTERISK_28638 caused a regression by incorrectly aborting early and overwriting the status on certain calls. This was exhibited by certain technologies such as DAHDI, where DAHDI returns NULL for the request if a line is busy. This caused the BUSY condition to be incorrectly treated as CHANUNAVAIL because the DIALSTATUS was getting incorrectly overwritten and call handling was aborted early. This is fixed by instead checking if any valid peers have been specified, as opposed to checking the list size of successful requests. This is because the latter could be empty but this does not indicate any kind of problem. This restores the previous working behavior. ASTERISK-29989 #close Change-Id: I4d4b209b967816b1bc791534593ababa2b99bb88	3 years ago
Kevin Harwell	628661ca42	res_pjsip: allow TLS verification of wildcard cert-bearing servers Rightly the use of wildcards in certificates is disallowed in accordance with RFC5922. However, RFC2818 does make some allowances with regards to their use when using subject alt names with DNS name types. As such this patch creates a new setting for TLS transports called 'allow_wildcard_certs', which when it and 'verify_server' are both enabled allows DNS name types, as well as the common name that start with '.' to match as a wildcard. For instance: .example.com will match for: foo.example.com Partial matching is not allowed, e.g. f.example.com, foo..com, etc... And the starting wildcard only matches for a single level. For instance: *.example.com will NOT match for: foo.bar.example.com The new setting is disabled by default. ASTERISK-30072 #close Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4	3 years ago
Naveen Albert	c76a107cdf	say: Abort play loop if caller hangs up. If the caller has hung up, break out of the play loop so we don't try to play remaining files and fail to do so. ASTERISK-30075 #close Change-Id: I55e85be28ee90b48c0fe4ce20ac136a7dbb49f14	3 years ago
Naveen Albert	b1359bdfcf	pbx: Add helper function to execute applications. Finding an application and executing it if found is a common task throughout Asterisk. This adds a helper function around pbx_exec to do this, to eliminate redundant code and make it easier for modules to substitute variables and execute applications by name. ASTERISK-30061 #close Change-Id: Ifee4d2825df7545fb515d763d393065675140c84	3 years ago
Stanislav Abramenkov	cd85f4a2a7	pjsip: Upgrade bundled version to pjproject 2.12.1 More information: https://github.com/pjsip/pjproject/releases/tag/2.12.1 Pull request to third-party https://github.com/asterisk/third-party/pull/11 ASTERISK-30050 Change-Id: Icb4e86d4b85ef9b975355c91f3ed56a50b51c6bd	3 years ago
Asterisk Development Team	786577c05b	Update CHANGES and UPGRADE.txt for 16.27.0	3 years ago
Naveen Albert	b7107b4dd4	asterisk.c: Fix incompatibility warnings for remote console. A previous review fixing ASTERISK_22246 and ASTERISK_26582 got a couple of the options mixed up as to whether or not they are compatible with the remote console. This fixes those to the best of my knowledge. ASTERISK-30097 #close Change-Id: Id54166991aa79f04fb02699cc499bedda854253b	3 years ago
Naveen Albert	a57e36edfc	sig_analog: Fix broken three-way conferencing. Three-way calling for analog lines is currently broken. If party A is on a call with party B and initiates a three-way call to party C, the behavior differs depending on whether the call is conferenced prior to party C answering. The post-answer case is correct. However, if A flashes before C answers, then the next flash disconnects B rather than C, which is incorrect. This error occurs because the subs are not swapped in the misbehaving case. This is because the flash handler only swaps the subs if C has answered already, which is wrong. To fix this, we swap the subs regardless of whether C has answered or not when the call is conferenced. This ensures that C is disconnected on the next hook flash, rather than B as can happen currently. ASTERISK-30043 #close Change-Id: I96c5bf6c9b7eb2636136b716c677c82c079b6f06	3 years ago
Kevin Harwell	5fe8583611	ARI version: increase non-breaking number A recent change, ASTERISK_30027, now exposes a channel driver's unique id to ARI channel resources so bump the ARI version. Change-Id: Ic9da84b7b0eca6f3ff5b6a738b2be8362237951e	3 years ago
Naveen Albert	47a1d7b1a3	app_voicemail: Add option to prevent message deletion. Adds an option to VoiceMailMain that prevents the user from deleting messages during that application invocation. This can be useful for public or shared mailboxes, where some users should be able to listen to messages but not delete them. ASTERISK-30063 #close Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26	3 years ago
Naveen Albert	8935333364	res_parking: Add music on hold override option. An m option to Park and ParkAndAnnounce now allows specifying a music on hold class override. ASTERISK-30087 Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e	3 years ago
Trevor Peirce	6dff26971f	res_pjsip: Actually enable session timers when timers=always When a pjsip endpoint is defined with timers=always, this has been a functional noop. This patch correctly sets the feature bitmap to both enable support for session timers and to enable them even when the endpoint itself does not request or support timers. ASTERISK-29603 Reported-By: Ray Crumrine Change-Id: I8b5eeaa9ec7f50cc6d96dd34c2b4aa9c53fb5440	3 years ago
Naveen Albert	4f3c774246	xmldocs: Improve examples. Use example tags instead of regular para tags where possible. ASTERISK-30090 Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5	3 years ago
Naveen Albert	c1611e7d1b	res_pjsip_outbound_registration: Make max random delay configurable. Currently, PJSIP will randomly wait up to 10 seconds for each outbound registration's initial attempt. The reason for this is to avoid having all outbound registrations attempt to register simultaneously. This can create limitations with the test suite where we need to be able to receive inbound calls potentially within 10 seconds of starting up. For instance, we might register to another server and then try to receive a call through the registration, but if the registration hasn't happened yet, this will fail, and hence this inconsistent behavior can cause tests to fail. Ultimately, this requires a smaller random value because there may be no good reason to wait for up to 10 seconds in these circumstances. To address this, a new config option is introduced which makes this maximum delay configurable. This allows, for instance, this to be set to a very small value in test systems to ensure that registrations happen immediately without an unnecessary delay, and can be used more generally to control how "tight" the initial outbound registrations are. ASTERISK-29965 #close Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd	3 years ago
Alexei Gradinari	0350a05aea	res_pjsip_pubsub: delete scheduled notification on RLS update If there is scheduled notification, we must delete it to avoid using destroyed subscriptions. ASTERISK-29906 Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f	3 years ago
Alexei Gradinari	6a73f7aca8	res_pjsip_pubsub: XML sanitized RLS display name ASTERISK-29891 Change-Id: Ic8c9697e616446e06e6302653eae902aa23372ad	3 years ago
Christof Efkemann	58cf5d3912	app_sayunixtime: Use correct inflection for German time. In function ast_say_date_with_format_de(), take special care when the hour is one o'clock. In this case, the German number "eins" must be inflected to its neutrum form, "ein". This is achieved by playing "digits/1N" instead of "digits/1". Fixes both 12- and 24-hour formats. ASTERISK-30092 Change-Id: Ica9b80125c0b317e378d89c1ea786816e2635510	3 years ago
Naveen Albert	82cebaa023	chan_iax2: Prevent deadlock due to duplicate autoservice. If a switch is invoked using chan_iax2, deadlock can result because the PBX core is autoservicing the channel while chan_iax2 also then attempts to service it while waiting for the result of the switch. This removes servicing of the channel to prevent any conflicts. ASTERISK-30064 #close Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22	3 years ago
Naveen Albert	3b7bcbb6d5	res_calendar: Prevent assertion if event ends in past. res_calendar will trigger an assertion currently if the ending time is calculated to be in the past. Unlike the reminder and start times, however, there is currently no check to catch non-positive times and set them to 1. As a result, if we get a negative value by happenstance, this can cause a crash. To prevent the assertion from begin triggered, we now use the same logic as the reminder and start events to catch this issue before it can cause a problem. ASTERISK-29981 #close Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c	3 years ago
Naveen Albert	8ec9e58eb4	res_parking: Warn if out of bounds parking spot requested. Emits a warning if the user has requested a parking spot that is out of bounds for the requested parking lot. ASTERISK-30086 Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf	3 years ago
Naveen Albert	ea8d2ca17c	loader: Prevent deadlock using tab completion. If tab completion using ast_module_helper is attempted during startup, deadlock will ensue because the CLI will attempt to lock the module list while it is already locked by the loader. This causes deadlock because when the loader tries to acquire the CLI lock, they are blocked on each other. Waiting for startup to complete is not feasible because the CLI lock is acquired while waiting, so deadlock will ensure regardless of whether or not a lock on the module list is attempted. To prevent deadlock, we immediately abort if tab completion is attempted on the module list before Asterisk is fully booted. ASTERISK-30039 #close Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d	3 years ago
Maximilian Fridrich	9ae06885fc	chan_pjsip: Only set default audio stream on hold. When a PJSIP channel is set on hold or off hold, all streams were set on/off hold. This is not the desired behaviour and caused issues when there were multiple streams in the topology. Now, only the default audio stream is set on/off hold when a hold is indicated. ASTERISK-30051 Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71	3 years ago
Alexei Gradinari	3ec5eb5ae6	res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local" Identity Element URI and Target Element URI to the same value - the channel Caller Number. For Identity Element it's ok to set as Caller ID. But Local Target URI should be set as local URI. In this case the Local Target URI can be used for Directed Call Pickup by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup). Also XML sanitized Display names. ASTERISK-24601 Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343	3 years ago
Shloime Rosenblum	e8616a2701	res_agi: Evaluate dialplan functions and variables in agi exec if enabled Agi commnad exec can now evaluate dialplan functions and variables if variable AGIEXECFULL is set to yes. this can be useful when executing Playback or Read from agi. ASTERISK-30058 #close Change-Id: I669991f540496e7bddd096fec82b52c083036832	3 years ago
Moritz Fain	4f2bd069a4	ari: expose channel driver's unique id to ARI channel resource This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Sean Bright	95daff54ca	loader.c: Use portable printf conversion specifier for int64. ASTERISK-30060 #close Change-Id: I88d47a1488be2f39017b8d562f993f081844fcb8	3 years ago

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