asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	1cfd9216b7	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 month ago
George Joseph	bda1d32001	.github: Update Releaser to use SES email	2 months ago
George Joseph	de6aaa9623	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.	2 months ago
George Joseph	076423aa18	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	2 months ago
Naveen Albert	45ee5cda1c	sig_analog: Fix SEGV due to calling strcmp on NULL. Add an additional check to guard against the channel application being NULL. Resolves: #1380	2 months ago
Sven Kube	a0b36f8099	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.	2 months ago
Naveen Albert	286c104231	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	2 months ago
Jose Lopes	4b9a373ec5	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	2 months ago
Naveen Albert	e182531373	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.	2 months ago
George Joseph	bd1e723eca	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.	2 months ago
Naveen Albert	00b6ab22a0	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	2 months ago
Naveen Albert	d04f273955	test_res_prometheus: Fix compilation failure on Debian 13. curl_easy_setopt expects long types, so be explicit. Resolves: #1369	2 months ago
Naveen Albert	40e6f2b12c	func_frame_drop: Handle allocation failure properly. Handle allocation failure and simplify the allocation using asprintf. Resolves: #1366	2 months ago
Alexey Khabulyak	a321038cad	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	2 months ago
Naveen Albert	3ea3ec6f0a	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	2 months ago
Ben Ford	5843d02541	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	2 months ago
Alexey Khabulyak	4b766d24d0	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	2 months ago
Mike Bradeen	8dce57f9ad	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	2 months ago
Allan Nathanson	59d6d2addf	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	2 months ago
Sperl Viktor	0906282f25	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.	2 months ago
George Joseph	830dd4656c	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	2 months ago
George Joseph	581e4a23a8	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.	2 months ago
zhou_jiajian	9d4674d93d	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'	2 months ago
Naveen Albert	99b310a33f	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.	2 months ago
George Joseph	853a171078	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	2 months ago
ThatTotallyRealMyth	fe9909dbb8	safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files. UpgradeNote: The safe_asterisk script now checks that, if it was run by the root user, the /etc/asterisk/startup.d directory and all the files it contains are owned by root. If the checks fail, safe_asterisk will exit with an error and Asterisk will not be started. Additionally, the default logging destination is now stderr instead of tty "9" which probably won't exist in modern systems. Resolves: #GHSA-v9q8-9j8m-5xwp	2 months ago
George Joseph	d1a1945967	options: Change ast_options from ast_flags to ast_flags64. DeveloperNote: The 32-bit ast_options has no room left to accomodate new options and so has been converted to an ast_flags64 structure. All internal references to ast_options have been updated to use the 64-bit flag manipulation macros. External module references to the 32-bit ast_options should continue to work on little-endian systems because the least-significant bytes of a 64 bit integer will be in the same location as a 32-bit integer. Because that's not the case on big-endian systems, we've swapped the bytes in the flags manupulation macros on big-endian systems so external modules should still work however you are encouraged to test.	2 months ago
Alexei Gradinari	9e3ea1c8ac	res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA) This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly falls back to the next configured backend when the current one returns SQL_NO_DATA (i.e., no record found). This is a logical error and performance risk in multi-backend configurations. Solution: Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker. ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found. Core engine stops iterating on this marker, avoiding unnecessary fallback. Notes: Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly. This patch only covers ODBC. Fixes: #1305	2 months ago
Sven Kube	0ba07ac7c3	resource_channels.c: Don't call ast_channel_get_by_name on empty optional arguments `ast_ari_channels_create` and `ast_ari_channels_dial` called the `ast_channel_get_by_name` function with optional arguments. Since `8f1982c4d6`, this function logs an error for empty channel names. This commit adds checks for empty optional arguments that are used to call `ast_channel_get_by_name` to prevent these error logs.	2 months ago
Naveen Albert	74e002277c	app_agent_pool: Remove documentation for removed option. The already-deprecated "password" option for the AGENT function was removed in commit `d43b17a872` for Asterisk 12, but the documentation for it wasn't removed then. Resolves: #1321	2 months ago
Tinet-mucw	4442e2dce1	pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper should return directly. Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by pbx_extension_helper this info is changed. This will cause the current dialplan location to be executed twice. In other words, the Redirect action does not take effect. Resolves: #1315	3 months ago
Sperl Viktor	3403da9c01	res_agi: Increase AGI command buffer size from 2K to 8K Fixes: #1317	3 months ago
Naveen Albert	9c55e72b67	ast_tls_cert: Make certificate validity configurable. Currently, the ast_tls_cert script is hardcoded to produce certificates with a validity of 365 days, which is not generally desirable for self- signed certificates. Make this parameter configurable. Resolves: #1307	3 months ago
George Joseph	a4aab9d39f	cdr.c: Set tenantid from party_a->base instead of chan->base. The CDR tenantid was being set in cdr_object_alloc from the channel->base snapshot. Since this happens at channel creation before the dialplan is even reached, calls to `CHANNEL(tenantid)=<something>` in the dialplan were being ignored. Instead we now take tenantid from party_a when cdr_object_create_public_records() is called which is after the call has ended and all channel snapshots rebuilt. This is exactly how accountcode and amaflags, which can also be set in tha dialplpan, are handled. Resolves: #1259	3 months ago
George Joseph	f459a57394	.github: Reduce number of inputs to Releaser to 10. The max number of inputs supported by GitHub is 10 so is_security and is_hotfix were factored into a single choice entry.	3 months ago
George Joseph	cc74f2f28c	.github: Add skip-cherry-pick and skip-test-builds to Releaser.	3 months ago
George Joseph	8a602dbfee	app_mixmonitor: Update the documentation concerning the "D" option. When using the "D" option to output interleaved audio, the file extension must be ".raw". That info wasn't being properly rendered in the markdown and HTML on the documentation site. The XML was updated to move the note in the option section to a warning in the description. Resolves: #1269	3 months ago
Naveen Albert	a45e6d6fd8	sig_analog: Properly handle STP, ST2P, and ST3P for fgccamamf. Previously, we were only using # (ST) as a terminator, and not handling A (STP), B (ST2P), or C (ST3P), which erroneously led to it being treated as part of the dialed number. Parse any of these as the start digit. Resolves: #1301	3 months ago
kodokaii	38d0909669	chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from the queue but does not reset the frame_queue_length counter. As a result, the driver incorrectly thinks the queue is full after flushing, which prevents new multimedia frames from being sent, especially after multiple flush commands. This fix sets frame_queue_length to 0 after flushing, ensuring the queue state is consistent with its actual content. Fixes: #1304	3 months ago
George Joseph	63e88f5bc1	channelstorage_cpp_map_name_id: Fix callback returning non-matching channels. When the callback() API was invoked but no channel passed the test, callback would return the last channel tested instead of NULL. It now correctly returns NULL when no channel matches. Resolves: #1288	3 months ago
Martin Tomec	5946bc6363	chan_pjsip.c: Change SSRC after media source change When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly. To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream. This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip. Resolves: #927	3 months ago
George Joseph	07fd3af897	Media over Websocket Channel Driver * Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket.	3 months ago
George Joseph	bf1e0a213f	channelstorage: Rename callbacks that conflict with DEBUG_FD_LEAKS. DEBUG_FD_LEAKS replaces calls to "open" and "close" with functions that keep track of file descriptors, even when those calls are actually callbacks defined in structures like ast_channelstorage_instance->open and don't touch file descriptors. This causes compilation failures. Those callbacks have been renamed to "open_instance" and "close_instance" respectively. Resolves: #1287	3 months ago
Stanislav Abramenkov	877e5ef770	bundled_pjproject: Avoid deadlock between transport and transaction Backport patch from upstream * Avoid deadlock between transport and transaction https://github.com/pjsip/pjproject/commit/edde06f261ac Issue described in https://github.com/pjsip/pjproject/issues/4442	3 months ago
Michal Hajek	a2ae527d60	audiohook.c: Improve frame pairing logic to avoid MixMonitor breakage with mixed codecs This patch adjusts the read/write synchronization logic in audiohook_read_frame_both() to better handle calls where participants use different codecs or sample sizes (e.g., alaw vs G.722). The previous hard threshold of 2 * samples caused MixMonitor recordings to break or stutter when frames were not aligned between both directions. The new logic uses a more tolerant limit (1.5 * samples), which prevents audio tearing without causing excessive buffer overruns. This fix specifically addresses issues with MixMonitor when recording directly on a channel in a bridge using mixed codecs. Reported-by: Michal Hajek <michal.hajek@daktela.com> Resolves: #1276 Resolves: #1279	3 months ago
Sean Bright	7d5280c16e	channelstorage_makeopts.xml: Remove errant XML character. Resolves: #1282	3 months ago
mkmer	55f4294e82	utils.h: Add rounding to float conversion to int. Quote from an audio engineer NR9V: There is a minor issue of a small amount of crossover distortion though as a result of `ast_slinear_saturated_multiply_float()` not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion. Resolves: #1176	3 months ago
Tinet-mucw	88f653c12d	pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return -1. Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by GotoIf this info is changed. that will causes confusion in dialplan execution. Resolves: #1273	3 months ago
Sean Bright	7feb676409	res_musiconhold.c: Ensure we're always locked around music state access.	3 months ago
Sean Bright	def223ffd8	res_musiconhold.c: Annotate when the channel is locked.	3 months ago

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34162 Commits (1cfd9216b7a576f72c5d6d16c85fcea00a2f7eab) All Branches Search

34162 Commits (1cfd9216b7a576f72c5d6d16c85fcea00a2f7eab)

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