asterisk

Commit Graph

Author	SHA1	Message	Date
phoneben	b502b7cf55	app_queue: Add NULL pointer checks in app_queue Add NULL check for word_list before calling word_in_list() Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel() Resolves: #1425	10 months ago
Sean Bright	a4d638fa02	app_externalivr: Prevent out-of-bounds read during argument processing. Resolves: #1422	10 months ago
Naveen Albert	930f5578e5	chan_dahdi: Add DAHDI_CHANNEL function. Add a dialplan function that can be used to get/set properties of DAHDI channels (as opposed to Asterisk channels). This exposes properties that were not previously available, allowing for certain operations to now be performed in the dialplan. Resolves: #1455 UserNote: The DAHDI_CHANNEL function allows for getting/setting certain properties about DAHDI channels from the dialplan.	10 months ago
Joshua C. Colp	b5729635d5	taskpool: Update versions for taskpool stasis options.	10 months ago
Joshua C. Colp	37800188ec	taskpool: Add taskpool API, switch Stasis to using it. This change introduces a new API called taskpool. This is a pool of taskprocessors. It provides the following functionality: 1. Task pushing to a pool of taskprocessors 2. Synchronous tasks 3. Serializers for execution ordering of tasks 4. Growing/shrinking of number of taskprocessors in pool This functionality already exists through the combination of threadpool+taskprocessors but through investigating I determined that this carries substantial overhead for short to medium duration tasks. The threadpool uses a single queue of work, and for management of threads it involves additional tasks. I wrote taskpool to eliminate the extra overhead and management as much as possible. Instead of a single queue of work each taskprocessor has its own queue and at push time a selector chooses the taskprocessor to queue the task to. Each taskprocessor also has its own thread like normal. This spreads out the tasks immediately and reduces contention on shared resources. Using the included efficiency tests the number of tasks that can be executed per second in a taskpool is 6-12 times more than an equivalent threadpool+taskprocessor setup. Stasis has been moved over to using this new API as it is a heavy consumer of threadpool+taskprocessors and produces a lot of tasks. UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for stasis. DeveloperNote: The taskpool API has been added for common usage of a pool of taskprocessors. It is suggested to use this API instead of the threadpool+taskprocessor approach.	10 months ago
Naveen Albert	2f4e432ad4	app_adsiprog: Fix possible NULL dereference. get_token can return NULL, but process_token uses this result without checking for NULL; as elsewhere, check for a NULL result to avoid possible NULL dereference. Resolves: #1419	10 months ago
Nathan Monfils	71730d7024	manager.c: Fix presencestate object leak ast_presence_state allocates subtype and message. We straightforwardly need to clean those up.	10 months ago
Sean Bright	19e9a93e9d	audiohook.c: Ensure correct AO2 reference is dereffed. Part of #1440.	10 months ago
Naveen Albert	524ca3b785	res_cliexec: Remove unnecessary casts to char*. Resolves: #1436	10 months ago
Ben Ford	1b9df386be	rtp_engine.c: Add exception for comfort noise payload. In a previous commit, a change was made to ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample rates. This ended up returning an invalid payload int for comfort noise. A check has been added that returns early if the payload is in fact supposed to be comfort noise. Fixes: #1340	10 months ago
Naveen Albert	fa9f322afa	pbx_variables.c: Create real channel for "dialplan eval function". "dialplan eval function" has been using a dummy channel for function evaluation, much like many of the unit tests. However, sometimes, this can cause issues for functions that are not expecting dummy channels. As an example, ast_channel_tech(chan) is NULL on such channels, and ast_channel_tech(chan)->type consequently results in a NULL dereference. Normally, functions do not worry about this since channels executing dialplan aren't dummy channels. While some functions are better about checking for these sorts of edge cases, use a real channel with a dummy technology to make this CLI command inherently safe for any dialplan function that could be evaluated from the CLI. Resolves: #1434	10 months ago
Asterisk Development Team	3bb594c9ca	Update for 23.0.0	11 months ago
Asterisk Development Team	6249324f40	Update for 23.0.0-rc2	12 months ago
George Joseph	86e3ed9620	res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used. In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when AST_RTP_INSTANCE_RTCP_MUX is set. Resolves: #1474	12 months ago
George Joseph	546cefb305	chan_websocket: Fix codec validation and add passthrough option. * Fixed an issue in webchan_write() where we weren't detecting equivalent codecs properly. * Added the "p" dialstring option that puts the channel driver in "passthrough" mode where it will not attempt to re-frame or re-time media coming in over the websocket from the remote app. This can be used for any codec but MUST be used for codecs that use packet headers or whose data stream can't be broken up on arbitrary byte boundaries. In this case, the remote app is fully responsible for correctly framing and timing media sent to Asterisk and the MEDIA text commands that could be sent over the websocket are disabled. Currently, passthrough mode is automatically set for the opus, speex and g729 codecs. * Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to ensure proper translation paths are set up when switching between native frames and slin silence frames. This fixes an issue with codec errors when transcode_via_sln=yes. Resolves: #1462	12 months ago
George Joseph	2b562c9531	res_ari: Ensure outbound websocket config has a websocket_client_id. Added a check to outbound_websocket_apply() that makes sure an outbound websocket config object in ari.conf has a websocket_client_id parameter. Resolves: #1457	12 months ago
Asterisk Development Team	de44952955	Update for 23.0.0-rc1	1 year ago
Joe Garlick	88d61c1d66	chan_websocket.c: Add DTMF messages Added DTMF messages to the chan_websocket feature. When a user presses DTMF during a call over chan_websocket it will send a message like: "DTMF_END digit:1" Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70 (cherry picked from commit `219ec0141c`)	1 year ago
Igor Goncharovsky	bff8884629	app_queue.c: Add new global 'log_unpause_on_reason_change' In many asterisk-based systems, the pause reason is used to separate pauses by type,and logically, changing the reason defines two intervals that should be accounted for separately. The introduction of a new option allows me to separate the intervals of operator inactivity in the log by the event of unpausing. UserNote: Add new global option 'log_unpause_on_reason_change' that is default disabled. When enabled cause addition of UNPAUSE event on every re-PAUSE with reason changed. (cherry picked from commit `c982efaceb`)	1 year ago
Igor Goncharovsky	c6527945c7	app_waitforsilence.c: Use milliseconds to calculate timeout time The functions WaitForNoise() and WaitForSilence() use the time() functions to calculate elapsed time, which causes the timer to fire on a whole second boundary, and the actual function execution time to fire the timer may be 1 second less than expected. This fix replaces time() with ast_tvnow(). Fixes: #1401 (cherry picked from commit `648b60f0f0`)	1 year ago
Ben Ford	fddda2b363	res_rtp_asterisk: Don't send RTP before DTLS has negotiated. There was no check in __rtp_sendto that prevented Asterisk from sending RTP before DTLS had finished negotiating. This patch adds logic to do so. Fixes: #1260 (cherry picked from commit `fb1e971b61`)	1 year ago
Artem Umerov	8ae9d59727	Fix missing ast_test_flag64 in extconf.c Fix missing ast_test_flag64 after `43bf8a4ded` (cherry picked from commit `387aad4348`)	1 year ago
Naveen Albert	98682d2fd5	pbx_builtins: Allow custom tone for WaitExten. Currently, the 'd' option will play dial tone while waiting for digits. Allow it to accept an argument for any tone from indications.conf. Resolves: #1396 UserNote: The tone used while waiting for digits in WaitExten can now be overridden by specifying an argument for the 'd' option. (cherry picked from commit `171d5e996c`)	1 year ago
Naveen Albert	9fcc6270be	res_tonedetect: Add option for TONE_DETECT detection to auto stop. One of the problems with TONE_DETECT as it was originally written is that if a tone is detected multiple times, it can trigger the redirect logic multiple times as well. For example, if we do an async goto in the dialplan after detecting a tone, because the detector is still active until explicitly disabled, if we detect the tone again, we will branch again and start executing that dialplan a second time. This is rarely ever desired behavior, and can happen if the detector is not removed quickly enough. Add a new option, 'e', which automatically disables the detector once the desired number of matches have been heard. This eliminates the potential race condition where previously the detector would need to be disabled immediately, but doing so quickly enough was not guaranteed. This also allows match criteria to be retained longer if needed, so the detector does not need to be destroyed prematurely. Resolves: #1390 UserNote: The 'e' option for TONE_DETECT now allows detection to be disabled automatically once the desired number of matches have been fulfilled, which can help prevent race conditions in the dialplan, since TONE_DETECT does not need to be disabled after a hit. (cherry picked from commit `ca8ac80edb`)	1 year ago
Stuart Henderson	347788f354	app_queue: fix comparison for announce-position-only-up Numerically comparing that the current queue position is less than last_pos_said can only be done after at least one announcement has been made, otherwise last_pos_said is at the default (0). Fixes: #1386 (cherry picked from commit `cfcb245634`)	1 year ago
George Joseph	a8e7f3c396	res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL. In the highly-unlikely event that get_authorization_hdr() couldn't find an Authorization header in a request, trying to get the digest algorithm would cauase a SEGV. We now check that we have an auth header that matches the realm before trying to get the algorithm from it. Resolves: #GHSA-64qc-9x89-rx5j (cherry picked from commit `810d6ae665`)	1 year ago
Alexei Gradinari	144f04d733	sorcery: Prevent duplicate objects and ensure missing objects are created on update This patch resolves two issues in Sorcery objectset handling with multiple backends: 1. Prevent duplicate objects: When an object exists in more than one backend (e.g., a contact in both 'astdb' and 'realtime'), the objectset previously returned multiple instances of the same logical object. This caused logic failures in components like the PJSIP registrar, where duplicate contact entries led to overcounting and incorrect deletions, when max_contacts=1 and remove_existing=yes. This patch ensures only one instance of an object with a given key is added to the objectset, avoiding these duplicate-related side effects. 2. Ensure missing objects are created: When using multiple writable backends, a temporary backend failure can lead to objects missing permanently from that backend. Currently, .update() silently fails if the object is not present, and no .create() is attempted. This results in inconsistent state across backends (e.g. astdb vs. realtime). This patch introduces a new global option in sorcery.conf: [general] update_or_create_on_update_miss = yes\|no Default: no (preserves existing behavior). When enabled: if .update() fails with no data found, .create() is attempted in that backend. This ensures that objects missing due to temporary backend outages are re-synchronized once the backend is available again. Added a new CLI command: sorcery show settings Displays global Sorcery settings, including the current value of update_or_create_on_update_miss. Updated tests to validate both flag enabled/disabled behavior. Fixes: #1289 UserNote: Users relying on Sorcery multiple writable backends configurations (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes in sorcery.conf to ensure missing objects are recreated after temporary backend failures. Default behavior remains unchanged unless explicitly enabled. (cherry picked from commit `14ea4d3b39`)	1 year ago
Naveen Albert	7b0566f8f4	sig_analog: Skip Caller ID spill if usecallerid=no. If Caller ID is disabled for an FXS port, then we should not send any Caller ID spill on the line, as we have no Caller ID information that we can/should be sending. Resolves: #1394 (cherry picked from commit `54cab33a43`)	1 year ago
Naveen Albert	0d45d465bc	chan_dahdi: Fix erroneously persistent dialmode. It is possible to modify the dialmode setting in the chan_dahdi/sig_analog private using the CHANNEL function, to modify it during calls. However, it was not being reset between calls, meaning that if, for example, tone dialing was disabled, it would never work again unless explicitly enabled. This fixes the setting by pairing it with a "perm" version of the setting, as a few other features have, so that it can be reset to the permanent setting between calls. The documentation is also clarified to explain the interaction of this setting and the digitdetect setting more clearly. Resolves: #1378 (cherry picked from commit `576a29b988`)	1 year ago
George Joseph	df3f2066d6	.github: Update Releaser to use SES email (cherry picked from commit `f44d1e480e`)	1 year ago
George Joseph	58621854fc	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `core show application Dial` that shows how to use it. Resolves: #1352 UserNote: A new WebSocket channel driver option `v` has been added to the Dial application that allows you to specify additional URI parameters on outgoing connections. Run `core show application Dial` from the Asterisk CLI to see how to use it. (cherry picked from commit `5d822d64ef`)	1 year ago
George Joseph	b3c4f4195f	chan_websocket: Fix buffer overrun when processing TEXT websocket frames. ast_websocket_read() receives data into a fixed 64K buffer then continually reallocates a final buffer that, after all continuation frames have been received, is the exact length of the data received and returns that to the caller. process_text_message() in chan_websocket was attempting to set a NULL terminator on the received payload assuming the payload buffer it received was the large 64K buffer. The assumption was incorrect so when it tried to set a NULL terminator on the payload, it could, depending on the state of the heap at the time, cause heap corruption. process_text_message() now allocates its own payload_len + 1 sized buffer, copies the payload received from ast_websocket_read() into it then NULL terminates it prevent the possibility of the overrun and corruption. Resolves: #1384 (cherry picked from commit `09802fbef2`)	1 year ago
Naveen Albert	c7ea11c35a	sig_analog: Fix SEGV due to calling strcmp on NULL. Add an additional check to guard against the channel application being NULL. Resolves: #1380 (cherry picked from commit `6f2167989a`)	1 year ago
Sven Kube	c97f4d844c	ARI: Add command to indicate progress to a channel Adds an ARI command to send a progress indication to a channel. DeveloperNote: A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel. (cherry picked from commit `b99f57e464`)	1 year ago
Naveen Albert	a59e0fee66	dsp.c: Improve debug logging in tone_detect(). The debug logging during DSP processing has always been kind of overwhelming and annoying to troubleshoot. Simplify and improve the logging in a few ways to aid DSP debugging: * If we had a DSP hit, don't also emit the previous debug message that was always logged. It is duplicated by the hit message, so this can reduce the number of debug messages during detection by 50%. * Include the hit count and required number of hits in the message so on partial detections can be more easily troubleshot. * Use debug level 9 for hits instead of 10, so we can focus on hits without all the noise from the per-frame debug message. * 1-index the hit count in the debug messages. On the first hit, it currently logs '0', just as when we are not detecting anything, which can be confusing. Resolves: #1375 (cherry picked from commit `dd9c71ed0b`)	1 year ago
Jose Lopes	1e84bcc8ec	res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart. After an asterisk restart, the deletion of ARI Devicestates didn't return error, but the devicestate was not deleted. Found a typo on populate_cache function that created wrong cache for device states. This bug caused wrong assumption that devicestate didn't exist, since it was not in cache, so deletion didn't returned error. Fixes: #1327 (cherry picked from commit `7c73bb235e`)	1 year ago
Naveen Albert	fd6b6bfb65	app_chanspy: Add option to not automatically answer channel. Add an option for ChanSpy and ExtenSpy to not answer the channel automatically. Most applications that auto-answer by default already have an option to disable this behavior if unwanted. Resolves: #1358 UserNote: ChanSpy and ExtenSpy can now be configured to not automatically answer the channel by using the 'N' option. (cherry picked from commit `5dd3ceaef5`)	1 year ago
George Joseph	8a4f745751	xmldoc.c: Fix rendering of CLI output. If you do a `core show application Dial`, you'll see it's kind of a mess. Indents are wrong is some places, examples are printed in black which makes them invisible on most terminals, and the lack of line breaks in some cases makes it hard to follow. * Fixed the rendering of examples so they are indented properly and changed the color so they can be seen. * There is now a line break before each option. * Options are now printed on their own line with all option content indented below them. Example from Dial before fixes: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest Example: Ringing FXS channel 4 with ring cadence 2 Example: Dial 555-1212 on channel 3 and require answer confirmation ... O([mode]): mode - With <mode> either not specified or set to '1', the originator hanging up will cause the phone to ring back immediately. - With <mode> set to '2', when the operator flashes the trunk, it will ring their phone back. Enables operator services mode. This option only works when bridging a DAHDI channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it will be ignored. When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party (the operator) hangs up. p: This option enables screening mode. This is basically Privacy mode without memory. ``` After: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest same => n,Dial(DAHDI/g1/5551212) Example: Ringing FXS channel 4 with ring cadence 2 same => n,Dial(DAHDI/4r2) Example: Dial 555-1212 on channel 3 and require answer confirmation same => n,Dial(DAHDI/3c/5551212) ... O([mode]): mode - With <mode> either not specified or set to '1', the originator hanging up will cause the phone to ring back immediately. With <mode> set to '2', when the operator flashes the trunk, it will ring their phone back. Enables operator services mode. This option only works when bridging a DAHDI channel to another DAHDI channel only. If specified on non-DAHDI interfaces, it will be ignored. When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party (the operator) hangs up. p: This option enables screening mode. This is basically Privacy mode without memory. ``` There are still things we can do to make this more readable but this is a start. (cherry picked from commit `59c9aa00b0`)	1 year ago
Naveen Albert	3d33980c02	func_frame_drop: Add debug messages for dropped frames. Add debug messages in scenarios where frames that are usually processed are dropped or skipped. Resolves: #1371 (cherry picked from commit `1ddf63d83f`)	1 year ago
Naveen Albert	e2525a4945	test_res_prometheus: Fix compilation failure on Debian 13. curl_easy_setopt expects long types, so be explicit. Resolves: #1369 (cherry picked from commit `3036fb7887`)	1 year ago
Naveen Albert	29efce1117	func_frame_drop: Handle allocation failure properly. Handle allocation failure and simplify the allocation using asprintf. Resolves: #1366 (cherry picked from commit `8ca6968317`)	1 year ago
Alexey Khabulyak	7d04ff2878	pbx_lua.c: segfault when pass null data to term_color function This can be reproduced under certain curcomstences. For example: call app.playback from lua with invalid data: app.playback({}). pbx_lua.c will try to get data for this playback using lua_tostring function. This function returs NULL for everything but strings and numbers. Then, it calls term_color with NULL data. term_color function can call(if we don't use vt100 compat term) ast_copy_string with NULL inbuf which cause segfault. bt example: ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0) at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412 Resolves: https://github.com/asterisk/asterisk/issues/1363 (cherry picked from commit `b098f2aa0f`)	1 year ago
Naveen Albert	2612ba32fc	bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel. If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed to answer when another channel bridges to it using Bridge(), and this is checked when ast_bridge_call* is called. However, another path exists (bridge_exec -> ast_bridge_add_channel) where this variable was not checked and channels would be answered. We now check the variable there. Resolves: #401 Resolves: #1364 (cherry picked from commit `a23c467c82`)	1 year ago
Ben Ford	ca5335d90e	Update version for Asterisk 23	1 year ago
Alexey Khabulyak	b0421fc87c	app_dial.c: Moved channel lock to prevent deadlock It's reproducible with pbx_lua, not regular dialplan. deadlock description: 1. asterisk locks a channel 2. calls function onedigit_goto 3. calls ast_goto_if_exists funciton 4. checks ast_exists_extension -> pbx_extension_helper 5. pbx_extension_helper calls pbx_find_extension 6. Then asterisk starts autoservice in a new thread 7. autoservice run tries to lock the channel again Because our channel is locked already, autoservice can't lock. Autoservice can't lock -> autoservice stop is waiting forever. onedigit_goto waits for autoservice stop. Resolves: https://github.com/asterisk/asterisk/issues/1335	1 year ago
Mike Bradeen	dce107234a	res_pjsip_diversion: resolve race condition between Diversion header processing and redirect Based on the firing order of the PJSIP call-backs on a redirect, it was possible for the Diversion header to not be included in the outgoing 181 response to the UAC and the INVITE to the UAS. This change moves the Diversion header processing to an earlier PJSIP callback while also preventing the corresponding update that can cause a duplicate 181 response when processing the header at that time. Resolves: #1349	1 year ago
Allan Nathanson	31571a5079	file.c: with "sounds_search_custom_dir = yes", search "custom" directory With `sounds_search_custom_dir = yes`, we are supposed to search for sounds in the `AST_DATA_DIR/sounds/custom` directory before searching the normal directories. Unfortunately, a recent change (https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in the "custom" directory not being searched. This change restores this expected behavior. Resolves: #1353	1 year ago
Sperl Viktor	c1f24b74d7	cel: Add STREAM_BEGIN, STREAM_END and DTMF event types. Fixes: #1280 UserNote: Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types in cel.conf will log media files and music on hold played to each channel. The STREAM_BEGIN event's extra field will contain a JSON with the file details (path, format and language), or the class name, in case of music on hold is played. The DTMF event's extra field will contain a JSON with the digit and the duration in milliseconds.	1 year ago
Naveen Albert	613d645d04	func_dialplan: Remove deprecated/redundant function. Remove VALID_EXTEN, which was deprecated/superseded by DIALPLAN_EXISTS in Asterisk 11 (commit `8017b65bb9`), as DIALPLAN_EXISTS does the same thing and is more flexible. Resolves: #1347 UpgradeNote: The deprecated VALID_EXTEN function has been removed. Use DIALPLAN_EXISTS instead.	1 year ago
Naveen Albert	c8f844a900	logger.c: Remove deprecated/redundant configuration option. Remove the deprecated 'rotatetimestamp' config option, as this was deprecated by 'rotatestrategy' in 1.6 by commit `f5a14167f3`. Resolves: #1345 UpgradeNote: The deprecated rotatetimestamp option has been removed. Use rotatestrategy instead.	1 year ago

1 2 3 4 5 ...

34218 Commits (b502b7cf55da5d8fd5817dbce291353cd4591b60) All Branches Search

34218 Commits (b502b7cf55da5d8fd5817dbce291353cd4591b60)

All Branches