asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	a11885989c	app_voicemail: Disable ADSI if unavailable. If ADSI is available on a channel, app_voicemail will repeatedly try to use ADSI, even if there is no CPE that supports it. This leads to many unnecessary delays during the session. If ADSI is available but ADSI setup fails, we now disable it to prevent further attempts to use ADSI during the session. Resolves: #354 (cherry picked from commit `75620616f4`)	2 years ago
Eduardo	d8238d0e15	codec_builtin: Use multiples of 20 for maximum_ms Some providers require a multiple of 20 for the maxptime or fail to complete calls, e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used. Resolves: #260 (cherry picked from commit `ca1ed84820`)	2 years ago
George Joseph	75bb76528a	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing. (cherry picked from commit `e1050b4add`)	2 years ago
George Joseph	fe1bca6a72	asterisk.c: Use the euid's home directory to read/write cli history The CLI .asterisk_history file is read from/written to the directory specified by the HOME environment variable. If the root user starts asterisk with the -U/-G options, or with runuser/rungroup set in asterisk.conf, the asterisk process is started as root but then it calls setuid/setgid to set the new user/group. This does NOT reset the HOME environment variable to the new user's home directory though so it's still left as "/root". In this case, the new user will almost certainly NOT have access to read from or write to the history file. * Added function process_histfile() which calls getpwuid(geteuid()) and uses pw->dir as the home directory instead of the HOME environment variable. * ast_el_read_default_histfile() and ast_el_write_default_histfile() have been modified to use the new process_histfile() function. Resolves: #337 (cherry picked from commit `fc516f5781`)	2 years ago
Tinet-mucw	aadf9d920a	res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes. From the gdb information, ast_websocket_read reads a message successfully, then transport_read is called in the serializer. During execution of pjsip_transport_down, ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop. After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages. This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue. In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop. Resolves: asterisk#299 (cherry picked from commit `a38add11e6`)	2 years ago
Mike Bradeen	f9efc9c681	cel: add publish user event helper Add a wrapper function around ast_cel_publish_event that packs event and extras into a blob before publishing Resolves:#330 (cherry picked from commit `4592f97c36`)	2 years ago
Naveen Albert	d01b047de3	chan_console: Fix deadlock caused by unclean thread exit. To terminate a console channel, stop_stream causes pthread_cancel to make stream_monitor exit. However, commit `5b8fea93d1` added locking to this function which results in deadlock due to the stream_monitor thread being killed while it's holding the pvt lock. To resolve this, a flag is now set and read to indicate abort, so the use of pthread_cancel and pthread_kill can be avoided altogether. Resolves: #308 (cherry picked from commit `3b027d1e47`)	2 years ago
George Joseph	dadbaed6f5	file.c: Add ability to search custom dir for sounds To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory. (cherry picked from commit `0e0f99db1d`)	2 years ago
Naveen Albert	69338381ea	chan_iax2: Improve authentication debugging. Improves and adds some logging to make it easier for users to debug authentication issues. Resolves: #286 (cherry picked from commit `f93138bcad`)	2 years ago
Vitezslav Novy	e1231fa4ac	res_rtp_asterisk: fix wrong counter management in ioqueue objects In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased which prevents unused ICE TURN threads from being removed. Resolves: #301 (cherry picked from commit `56244a7371`)	2 years ago
George Joseph	58c8d60f4b	res_pjsip_pubsub: Add body_type to test_handler for unit tests The ast_sip_subscription_handler "test_handler" used for the unit tests didn't set "body_type" so the NULL value was causing a SEGV in build_subscription_tree(). It's now set to "". Resolves: #335 (cherry picked from commit `71d75373f9`)	2 years ago
George Joseph	e2c3fd8d41	make_buildopts_h, et. al. Allow adding all cflags to buildopts.h The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options. (cherry picked from commit `42f82d55b0`)	2 years ago
Mike Bradeen	4799c0eda3	func_periodic_hook: Add hangup step to avoid timeout func_periodic_hook does not hangup after playback, relying on hangup which keeps the channel alive longer than necessary. Resolves: #325 (cherry picked from commit `2265eafb3a`)	2 years ago
Sean Bright	fd7a35fad8	res_stasis_recording.c: Save recording state when unmuted. Fixes #322 (cherry picked from commit `b952a8c38d`)	2 years ago
Mike Bradeen	2fb8dbc679	res_speech_aeap: check for null format on response * Fixed issue in res_speech_aeap when unable to provide an input format to check against. (cherry picked from commit `3759d034cc`)	2 years ago
George Joseph	e4edc9c75d	func_periodic_hook: Don't truncate channel name func_periodic_hook was truncating long channel names which causes issues when you need to run other dialplan functions/apps on the channel. Resolves: #319 (cherry picked from commit `cff637a24f`)	2 years ago
George Joseph	569dc4fb43	safe_asterisk: Change directory permissions to 755 If the safe_asterisk script detects that the /var/lib/asterisk directory doesn't exist, it now creates it with 755 permissions instead of 770. safe_asterisk needing to create that directory should be extremely rare though because it's normally created by 'make install' which already sets the permissions to 755. Resolves: #316 (cherry picked from commit `6bd9f9ae81`)	2 years ago
Maximilian Fridrich	3829c94d71	chan_rtp: Implement RTP glue for UnicastRTP channels Resolves: #298 UserNote: The dial string option 'g' was added to the UnicastRTP channel which enables RTP glue and therefore native RTP bridges with those channels. (cherry picked from commit `1af2ae177c`)	2 years ago
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`)	2 years 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`)	2 years ago
George Joseph	2f5dc8985a	Restore CHANGES and UPGRADE.txt to allow cherry-picks to work	2 years ago
Asterisk Development Team	71ddb39bd9	Update for 21.0.2	2 years 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	2 years ago
Asterisk Development Team	beba569755	Update for 21.0.1	2 years 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.	2 years 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.	2 years 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	2 years 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.	2 years 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

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