asterisk

Commit Graph

Author	SHA1	Message	Date
Jaco Kroon	9eeee41fc5	app_queue: periodic announcement configurable start time. This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. UserNote: Introduce a new queue configuration option called 'periodic-announce-startdelay' which will vary the normal (historic) behavior of starting the periodic announcement cycle at periodic-announce-frequency seconds after entering the queue to start the periodic announcement cycle at period-announce-startdelay seconds after joining the queue. The default behavior if this config option is not set remains unchanged. Signed-off-by: Jaco Kroon <jaco@uls.co.za> (cherry picked from commit `130c3ab792`)	1 year ago
Joshua C. Colp	73179928e6	variables: Add additional variable dialplan functions. Using the Set dialplan application does not actually delete channel or global variables. Instead the variables are set to an empty value. This change adds two dialplan functions, GLOBAL_DELETE and DELETE which can be used to delete global and channel variables instead of just setting them to empty. There is also no ability within the dialplan to determine if a global or channel variable has actually been set or not. This change also adds two dialplan functions, GLOBAL_EXISTS and VARIABLE_EXISTS which can be used to determine if a global or channel variable has been set or not. Resolves: #289 UserNote: Four new dialplan functions have been added. GLOBAL_DELETE and DELETE have been added which allows the deletion of global and channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been added which checks whether a global or channel variable has been set. (cherry picked from commit `182ad6926b`)	1 year ago
George Joseph	2f5dc8985a	Restore CHANGES and UPGRADE.txt to allow cherry-picks to work	1 year ago
Asterisk Development Team	71ddb39bd9	Update for 21.0.2	1 year ago
George Joseph	a49bb17dbb	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
Asterisk Development Team	beba569755	Update for 21.0.1	1 year ago
Gitea	b9594cc08a	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	8043d060e3	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	be5e9c568d	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	1c4d6d3af1	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
Asterisk Development Team	12da95e53f	Update for 21.0.0	2 years ago
Asterisk Development Team	f53f391889	Update for 21.0.0-rc1	2 years ago
George Joseph	bbbc9a8540	ari-stubs: Fix more local anchor references Also allow CreateDocs job to be run manually with default branches. (cherry picked from commit `c457784bd0`)	2 years ago
George Joseph	a0fc6c3ab0	ari-stubs: Fix more local anchor references Also allow CreateDocs job to be run manually with default branches. (cherry picked from commit `b68006f3a1`)	2 years ago
George Joseph	1108db21ef	ari-stubs: Fix broken documentation anchors All of the links that reference page anchors with capital letters in the ids (#Something) have been changed to lower case to match the anchors that are generated by mkdocs. (cherry picked from commit `c4508accc6`)	2 years ago
Bastian Triller	28320a9cd8	res_pjsip_session: Send Session Interval too small response Handle session interval lower than endpoint's configured minimum timer when sending first answer. Timer setting is checked during this step and needs to handled appropriately. Before this change, no response was sent at all. After this change a response with 422 Session Interval too small is sent to UAC. (cherry picked from commit `9284dca636`)	2 years ago
George Joseph	4b11eb23c8	.github: Update workflow-application-token-action to v2 (cherry picked from commit `83964ede1f`)	2 years ago
Naveen Albert	3b7a0d617f	app_dial: Fix infinite loop when sending digits. If the called party hangs up while digits are being sent, -1 is returned to indicate so, but app_dial was not checking the return value, resulting in the hangup being lost and looping forever until the caller manually hangs up the channel. We now abort if digit sending fails. ASTERISK-29428 #close Resolves: #281 (cherry picked from commit `c032d38d3f`)	2 years ago
Mike Bradeen	5f7c8e75fd	app_voicemail: Fix for loop declarations Resolve for loop initial declarations added in cli changes. Resolves: #275 (cherry picked from commit `e9b269692f`)	2 years ago
George Joseph	c4e2c00553	alembic: Fix quoting of the 100rel column Add quoting around the ps_endpoints 100rel column in the ALTER statements. Although alembic doesn't complain when generating sql statements, postgresql does (rightly so). Resolves: #274 (cherry picked from commit `f8438714ca`)	2 years ago
Naveen Albert	d29a7c19d0	pbx.c: Fix gcc 12 compiler warning. Resolves: #277 (cherry picked from commit `9e6266e008`)	2 years ago
zhengsh	d15a5c547f	app_audiosocket: Fixed timeout with -1 to avoid busy loop. Resolves: asterisk#234 (cherry picked from commit `eab40e755e`)	2 years ago
George Joseph	62e5a2720a	download_externals: Fix a few version related issues * Fixed issue with the script not parsing the new tag format for certified releases. The format changed from certified/18.9-cert5 to certified-18.9-cert5. * Fixed issue where the asterisk version wasn't being considered when looking for cached versions. Resolves: #263 (cherry picked from commit `644ad2f9cd`)	2 years ago
Maximilian Fridrich	9cb77904e0	main/refer.c: Fix double free in refer_data_destructor + potential leak Resolves: #267 (cherry picked from commit `48e6a482c2`)	2 years ago
Naveen Albert	e2f0538dea	sig_analog: Add Called Subscriber Held capability. This adds support for Called Subscriber Held for FXS lines, which allows users to go on hook when receiving a call and resume the call later from another phone on the same line, without disconnecting the call. This is a convenience mechanism that most real PSTN telephone switches support. ASTERISK-30372 #close Resolves: #240 UserNote: Called Subscriber Held is now supported for analog FXS channels, using the calledsubscriberheld option. This allows a station user to go on hook when receiving an incoming call and resume from another phone on the same line by going on hook, without disconnecting the call. (cherry picked from commit `cd0bfe193f`)	2 years ago
Matthew Fredrickson	64f76bdc60	Revert "app_stack: Print proper exit location for PBXless channels." This reverts commit `617dad4cba`. apps/app_stack.c: Revert buggy gosub patch This seems to break the case when a predial macro calls a gosub. When the gosub calls return, the Return function outputs: app_stack.c:423 return_exec: Return without Gosub: stack is empty This returns -1 to the calling macro, which returns to app_dial and causes the call to hangup instead of proceeding with the macro that invoked the gosub. Resolves: #253 (cherry picked from commit `1d7ae8d227`)	2 years ago
Jason D. McCormick	41e4673eb0	install_prereq: Fix dependency install on aarch64. Fixes dependency solutions in install_prereq for Debian aarch64 platforms. install_prereq was attempting to forcibly install 32-bit armhf packages due to the aptitude search for dependencies. Resolves: #37 (cherry picked from commit `2f02095da8`)	2 years ago
MikeNaso	59618c3a34	res_pjsip.c: Set contact_user on incoming call local Contact header If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls. Resolves: #226 (cherry picked from commit `720813dc97`)	2 years ago
Sean Bright	7e8aadae5a	extconfig: Allow explicit DB result set ordering to be disabled. Added a new boolean configuration flag - `order_multi_row_results_by_initial_column` - to both res_pgsql.conf and res_config_odbc.conf that allows the administrator to disable the explicit `ORDER BY` that was previously being added to all generated SQL statements that returned multiple rows. Fixes: #179 (cherry picked from commit `1171beb7e4`)	2 years ago
George Joseph	f287e8fce4	rest-api: Run make ari-stubs An earlier cherry-pick that involved rest-api somehow didn't include a comment change in res/ari/resource_endpoints.h. This commit corrects that. No changes other than the comment. (cherry picked from commit `21b0522abd`)	2 years ago
Naveen Albert	c011914f70	res_pjsip_header_funcs: Make prefix argument optional. The documentation for PJSIP_HEADERS claims that prefix is optional, but in the code it is actually not. However, there is no inherent reason for this, as users may want to retrieve all header names, not just those beginning with a certain prefix. This makes the prefix optional for this function, simply fetching all header names if not specified. As a result, the documentation is now correct. Resolves: #230 UserNote: The prefix argument to PJSIP_HEADERS is now optional. If not specified, all header names will be returned. (cherry picked from commit `2179082eaf`)	2 years ago
George Joseph	e37cfa85bb	pjproject_bundled: Increase PJSIP_MAX_MODULE to 38 The default is 32 with 8 being used by pjproject itself. Recent commits have put us over the limit resulting in assertions in pjproject. Since this value is used in invites, dialogs, transports and subscriptions as well as the global pjproject endpoint, we don't want to increase it too much. Resolves: #255 (cherry picked from commit `d3ee0e6516`)	2 years ago
Joshua C. Colp	f549da4cf2	manager: Tolerate stasis messages with no channel snapshot. In some cases I have yet to determine some stasis messages may be created without a channel snapshot. This change adds some tolerance to this scenario, preventing a crash from occurring. (cherry picked from commit `a75035be55`)	2 years ago
George Joseph	f7ea380b04	Remove unneeded CHANGES and UPGRADE files (cherry picked from commit `ea7e719d71`)	2 years ago
George Joseph	68349125a8	Update version for Asterisk 21	2 years ago
Maximilian Fridrich	51a7b18038	core/ari/pjsip: Add refer mechanism This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.	2 years ago
Naveen Albert	d16046e41f	chan_dahdi: Allow autoreoriginating after hangup. Currently, if an FXS channel is still off hook when all calls on the line have hung up, the user is provided reorder tone until going back on hook again. In addition to not reflecting what most commercial switches actually do, it's very common for switches to automatically reoriginate for the user so that dial tone is provided without the user having to depress and release the hookswitch manually. This can increase convenience for users. This behavior is now supported for kewlstart FXS channels. It's supported only for kewlstart (FXOKS) mainly because the behavior doesn't make any sense for ground start channels, and loop start signalling doesn't provide the necessary DAHDI event that makes this easy to implement. Likely almost everyone is using FXOKS over FXOLS anyways since FXOLS is pretty useless these days. ASTERISK-30357 #close Resolves: #224 UserNote: The autoreoriginate setting now allows for kewlstart FXS channels to automatically reoriginate and provide dial tone to the user again after all calls on the line have cleared. This saves users from having to manually hang up and pick up the receiver again before making another call.	2 years ago
Joshua C. Colp	806515597e	audiohook: Unlock channel in mute if no audiohooks present. In the case where mute was called on a channel that had no audiohooks the code was not unlocking the channel, resulting in a deadlock. Resolves: #233	2 years ago
Naveen Albert	e1a1ae933b	sig_analog: Allow three-way flash to time out to silence. sig_analog allows users to flash and use the three-way dial tone as a primitive hold function, simply by never timing it out. Some systems allow this dial tone to time out to silence, so the user is not annoyed by a persistent dial tone. This option allows the dial tone to time out normally to silence. ASTERISK-30004 #close Resolves: #205 UserNote: The threewaysilenthold option now allows the three-way dial tone to time out to silence, rather than continuing forever.	2 years ago
Holger Hans Peter Freyther	f335da6b74	res_prometheus: Do not generate broken metrics In `8d6fdf9c3a` invisible bridges were skipped but that lead to producing metrics with no name and no help. Keep track of the number of metrics configured and then only emit these. Add a basic testcase that verifies that there is no '(NULL)' in the output. ASTERISK-30474	2 years ago
Sean Bright	c52b4ce11c	res_pjsip: Enable TLS v1.3 if present. Fixes #221 UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying PJSIP library. The bundled version of PJSIP supports TLS v1.3.	2 years ago
phoneben	83dd36ba13	func_cut: Add example to documentation. This adds an example to the XML documentation clarifying usage of the CUT function to address a common misusage.	2 years ago
Sean Bright	5c4cbeff87	extensions.conf.sample: Remove reference to missing context. `c3ff4648` removed the [iaxtel700] context but neglected to remove references to it. This commit addresses that and also removes iaxtel and freeworlddialup references from other config files.	2 years ago
Sean Bright	b22aabe64a	func_export: Use correct function argument as variable name. Fixes #208	2 years ago
Joshua C. Colp	6b6880072b	app_queue: Add support for applying caller priority change immediately. The app_queue module provides both an AMI action and a CLI command to change the priority of a caller in a queue. Up to now this change of priority has only been reflected to new callers into the queue. This change adds an "immediate" option to both the AMI action and CLI command which immediately applies the priority change respective to the other callers already in the queue. This can allow, for example, a caller to be placed at the head of the queue immediately if their priority is sufficient. Resolves: #202 UserNote: The 'queue priority caller' CLI command and 'QueueChangePriorityCaller' AMI action now have an 'immediate' argument which allows the caller priority change to be reflected immediately, causing the position of a caller to move within the queue depending on the priorities of the other callers.	2 years ago
George Joseph	d0e5f2f6be	.github: Fix cherry-pick reminder issues	2 years ago
George Joseph	8a864bcdba	app.h: Move declaration of ast_getdata_result before its first use The ast_app_getdata() and ast_app_getdata_terminator() declarations in app.h were changed recently to return enum ast_getdata_result (which is how they were defined in app.c). The existing declaration of ast_getdata_result in app.h was about 1000 lines after those functions however so under certain circumstances, a "use before declaration" error was thrown by the compiler. The declaration of the enum was therefore moved to before those functions. Resolves: #200	2 years ago
Sean Bright	508657879e	chan_iax2.c: Avoid crash with IAX2 switch support. A change made in `82cebaa0` did not properly handle the case when a channel was not provided, triggering a crash. ast_check_hangup(...) does not protect against NULL pointers. Fixes #180	2 years ago
Sean Bright	fe467d595c	res_geolocation: Ensure required 'location_info' is present. Fixes #189	2 years ago
Mike Bradeen	eef5a0b7bf	Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. Resolves: #181 UserNote: The following manager actions have been added VoicemailBoxSummary - Generate message list for a given mailbox VoicemailRemove - Remove a message from a mailbox folder VoicemailMove - Move a message from one folder to another within a mailbox VoicemailForward - Copy a message from one folder in one mailbox to another folder in another or the same mailbox.	2 years ago

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