asterisk

Commit Graph

Author	SHA1	Message	Date
Jaco Kroon	60b529ea00	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>	2 years ago
Joshua C. Colp	039bbddd4f	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.	2 years ago
George Joseph	c14849ac17	ari-stubs: Fix more local anchor references Also allow CreateDocs job to be run manually with default branches.	2 years ago
George Joseph	0771c94e23	ari-stubs: Fix more local anchor references Also allow CreateDocs job to be run manually with default branches.	2 years ago
George Joseph	540c8b2794	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.	2 years ago
Bastian Triller	98665a14b3	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.	2 years ago
George Joseph	29967e1eea	.github: Update workflow-application-token-action to v2	2 years ago
Naveen Albert	1ada3c0747	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	2 years ago
Mike Bradeen	62480fd91b	app_voicemail: Fix for loop declarations Resolve for loop initial declarations added in cli changes. Resolves: #275	2 years ago
George Joseph	39b8b76f7c	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	2 years ago
Naveen Albert	1241410bc3	pbx.c: Fix gcc 12 compiler warning. Resolves: #277	2 years ago
zhengsh	a2353f4803	app_audiosocket: Fixed timeout with -1 to avoid busy loop. Resolves: asterisk#234	2 years ago
George Joseph	79e533e15b	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	2 years ago
Maximilian Fridrich	0950d116af	main/refer.c: Fix double free in refer_data_destructor + potential leak Resolves: #267	2 years ago
Naveen Albert	333858fb70	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.	2 years ago
Matthew Fredrickson	ec4e0340ce	app_macro: Fix locking around datastore access app_macro sometimes would crash due to datastore list corruption on the channel because of lack of locking around find and create process for the macro datastore. This patch locks the channel lock prior to protect against this problem. Resolves: #265	2 years ago
Matthew Fredrickson	3f80f9ce0b	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	2 years ago
George Joseph	a70c39acf8	.github: Use generic releaser	2 years ago
Jason D. McCormick	ef59542668	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	2 years ago
MikeNaso	4fc07f6d4e	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	2 years ago
Sean Bright	0bb192831c	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	2 years ago
George Joseph	6209d47b44	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.	2 years ago
Naveen Albert	08e7d77fce	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.	2 years ago
George Joseph	cb7a95d465	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	2 years ago
Joshua C. Colp	9b607747ce	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.	2 years ago
Maximilian Fridrich	57f77e8218	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	c08c458fa2	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	674bb1c9fe	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	a670655b7a	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	9b93fb9f67	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	8e23f3e313	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	3e91137404	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	f389d5d4b5	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	d36d208beb	func_export: Use correct function argument as variable name. Fixes #208	2 years ago
Joshua C. Colp	71558e23d0	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	aa03860504	.github: Fix cherry-pick reminder issues	2 years ago
George Joseph	d54695914d	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
George Joseph	dd1940cacf	doc: Remove obsolete CHANGES-staging and UPGRADE-staging	2 years ago
Sean Bright	39912b9138	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	4881ec18d0	res_geolocation: Ensure required 'location_info' is present. Fixes #189	2 years ago
Mike Bradeen	74f62dbfe9	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
Mike Bradeen	6aff1c1065	app_voicemail: add CLI commands for message manipulation Adds CLI commands to allow move/remove/forward individual messages from a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a show mailbox, required to retrieve message ID's. Resolves: #170 UserNote: The following CLI commands have been added to app_voicemail voicemail show mailbox <mailbox> <context> Show contents of mailbox <mailbox>@<context> voicemail remove <mailbox> <context> <from_folder> <messageid> Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context> voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder> Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder> voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder> Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to mailbox <mailbox>@<context> <to_folder>	2 years ago
zhengsh	fe478ddc14	res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. From the gdb information, it was found that when calling __ast_free, the size of the allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb, it is found to be 1. Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid, which is outside the protection of the rtp_instance lock. However, ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses rtp->themssrc_valid within the protection of the rtp_instance lock. This can lead to the possibility that the value of rtp->themssrc_valid used in the call to ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used within ast_rtcp_generate_report(). Resolves: asterisk#63	2 years ago
George Joseph	f5cb01afaf	.github: Minor tweak to Asterisk Releaser	2 years ago
George Joseph	53eb5ceae9	.github: Suppress cherry-pick reminder for some situations In PROpenedOrUpdated, the cherry-pick reminder will now be suppressed if there are already valid 'cherry-pick-to' comments in the PR or the PR contained a 'cherry-pick-to: none' comment.	2 years ago
Naveen Albert	0d1ddb15db	sig_analog: Allow immediate fake ring to be suppressed. When immediate=yes on an FXS channel, sig_analog will start fake audible ringback that continues until the channel is answered. Even if it answers immediately, the ringback is still audible for a brief moment. This can be disruptive and unwanted behavior. This adds an option to disable this behavior, though the default behavior remains unchanged. ASTERISK-30003 #close Resolves: #118 UserNote: The immediatering option can now be set to no to suppress the fake audible ringback provided when immediate=yes on FXS channels.	2 years ago
Sean Bright	ab0d6083fc	apply_patches: Use globbing instead of file/sort. This accomplishes the same thing as a `find ... \| sort` but with the added benefit of clarity and avoiding a call to a subshell. Additionally drop the -s option from call to patch as it is not POSIX.	2 years ago
George Joseph	7a725e3159	apply_patches: Sort patch list before applying The apply_patches script wasn't sorting the list of patches in the "patches" directory before applying them. This left the list in an indeterminate order. In most cases, the list is actually sorted but rarely, they can be out of order and cause dependent patches to fail to apply. We now sort the list but the "sort" program wasn't in the configure scripts so we needed to add that and regenerate the scripts as well. Resolves: #193	2 years ago
Stanislav Abramenkov	e341a41578	pjsip: Upgrade bundled version to pjproject 2.13.1	2 years ago
George Joseph	378166230a	.github: Updates for AsteriskReleaser	2 years ago

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