34201 Commits (b15a745ef895f87e0c890edfcb689372dc84340b)

 

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Author	SHA1	Message	Date
Sven Kube	b15a745ef8	stasis_channels.c: Make protocol_id optional to enable blind transfer via ari ... When handling SIP transfers via ARI, there is no protocol_id in case of a blind transfer. Resolves: #1467	1 year ago
George Joseph	a7b392e869	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	1 year ago
Bastian Triller	eae91bc317	Fix some doxygen, typos and whitespace	1 year ago
Sven Kube	8b24a226de	stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create ... When handling SIP transfers via ARI, the `referred_by` field in `transfer_ari_state` may be null, since SIP REFER requests are not required to include a `Referred-By` header. Without this check, a null value caused the transfer to fail and triggered a NOTIFY with a 500 Internal Server Error.	1 year ago
George Joseph	b624b92656	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	1 year ago
phoneben	72032b9493	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	1 year ago
Sean Bright	76dbd455a0	app_externalivr: Prevent out-of-bounds read during argument processing. ... Resolves: #1422	1 year ago
Naveen Albert	83bda237d9	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.	1 year ago
George Joseph	b8b079f55f	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	1 year ago
Naveen Albert	ff8940967f	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	1 year ago
Nathan Monfils	85cf8fcae8	manager.c: Fix presencestate object leak ... ast_presence_state allocates subtype and message. We straightforwardly need to clean those up.	1 year ago
Sean Bright	c1f5c1abe5	audiohook.c: Ensure correct AO2 reference is dereffed. ... Part of #1440.	1 year ago
Naveen Albert	d297afcd72	res_cliexec: Remove unnecessary casts to char*. ... Resolves: #1436	1 year ago
Ben Ford	0a0d7e17e7	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	1 year ago
Naveen Albert	7cf3077ee9	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	1 year ago
Joe Garlick	8dbf144e16	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	1 year ago
Igor Goncharovsky	744e8d3938	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.	1 year ago
Igor Goncharovsky	2e95a334a5	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	1 year ago
Artem Umerov	c291f67847	Fix missing ast_test_flag64 in extconf.c ... Fix missing ast_test_flag64 after 43bf8a4ded	1 year ago
Naveen Albert	bf2565bbe0	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.	1 year ago
Naveen Albert	6606fe8efe	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.	1 year ago
Stuart Henderson	0baf09b455	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	1 year ago
George Joseph	89001d3156	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	1 year ago
Alexei Gradinari	82f5a45a9e	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.	1 year ago
Naveen Albert	c27b09e155	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	1 year ago
Naveen Albert	7e68659616	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	1 year ago
George Joseph	e3877360f6	.github: Update Releaser to use SES email	1 year ago
George Joseph	d4d7f2e6e4	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.	1 year ago
George Joseph	72d1b469fd	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	1 year ago
Naveen Albert	4a1a8987c2	sig_analog: Fix SEGV due to calling strcmp on NULL. ... Add an additional check to guard against the channel application being NULL. Resolves: #1380	1 year ago
Sven Kube	71b538e79f	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.	1 year ago
Naveen Albert	8bfa3be27f	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	1 year ago
Jose Lopes	8837723f7d	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	1 year ago
Naveen Albert	94afcffce9	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.	1 year ago
George Joseph	95d65bbe30	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.	1 year ago
Naveen Albert	a039bb58a4	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	1 year ago
Naveen Albert	d48bd100eb	test_res_prometheus: Fix compilation failure on Debian 13. ... curl_easy_setopt expects long types, so be explicit. Resolves: #1369	1 year ago
Naveen Albert	16f789c8f8	func_frame_drop: Handle allocation failure properly. ... Handle allocation failure and simplify the allocation using asprintf. Resolves: #1366	1 year ago
Alexey Khabulyak	9bfd4c2994	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	1 year ago
Naveen Albert	6bc336b0ca	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	1 year ago
Ben Ford	b5146839e6	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	1 year ago
Alexey Khabulyak	15f93f8cfa	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	9fa3bb031e	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	80fd650881	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	0391c3f36e	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
George Joseph	e50d0cf741	channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety. ... The fact that deleting an object from a map invalidates any iterator that happens to currently point to that object was overlooked in the initial implementation. Unfortunately, there's no way to detect that an iterator has been invalidated so the result was an occasional SEGV triggered by modules like app_chanspy that opens an iterator and can keep it open for a long period of time. The new implementation doesn't keep the underlying C++ iterator open across calls to ast_channel_iterator_next() and uses a read lock on the map to ensure that, even for the few microseconds we use the iterator, another thread can't delete a channel from under it. Even with this change, the iterators are still WAY faster than the ao2_legacy storage driver. Full details about the new implementation are located in the comments for iterator_next() in channelstorage_cpp_map_name_id.cc. Resolves: #1309	1 year ago
George Joseph	706a7e45f0	res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM ... UserNote: Options are now available in the menuselect "Resource Modules" category that allow you to enable the AES_192, AES_256 and AES_GCM cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support them but modern versions do. Previously, the only way to enable them was to set the CFLAGS environment variable when running ./configure. The default setting is to disable them preserving existing behavior.	1 year ago
zhou_jiajian	5549815df5	cdr: add CANCEL dispostion in CDR ... In the original implementation, both CANCEL and NO ANSWER states were consolidated under the NO ANSWER disposition. This patch introduces a separate CANCEL disposition, with an optional configuration switch to enable this new disposition. Resolves: #1323 UserNote: A new CDR option "canceldispositionenabled" has been added that when set to true, the NO ANSWER disposition will be split into two dispositions: CANCEL and NO ANSWER. The default value is 'no'	1 year ago
Naveen Albert	3707712bb7	func_curl: Allow auth methods to be set. ... Currently the CURL function only supports Basic Authentication, the default auth method in libcurl. Add an option that also allows enabling digest authentication. Resolves: #1332 UserNote: The httpauth field in CURLOPT now allows the authentication methods to be set.	1 year ago
George Joseph	a60b15d66c	res_stir_shaken: Test for missing semicolon in Identity header. ... ast_stir_shaken_vs_verify() now makes sure there's a semicolon in the Identity header to prevent a possible segfault. Resolves: #GHSA-mrq5-74j5-f5cr	1 year ago