34161 Commits (9df2ba4eef3dc3029ae41aeb7722273caa5493be)

 

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Author	SHA1	Message	Date
Ben Ford	9df2ba4eef	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	6 months ago
Alexey Khabulyak	09a3c8df70	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	6 months ago
Mike Bradeen	4705601f17	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	6 months ago
Allan Nathanson	3ea4289a9a	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	6 months ago
Sperl Viktor	f516f07fd2	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.	6 months ago
George Joseph	06622f310b	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	6 months ago
George Joseph	ae078af893	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.	6 months ago
zhou_jiajian	01661398a3	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'	6 months ago
Naveen Albert	06e05a41c3	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.	6 months ago
George Joseph	e7ae1f7cf1	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	6 months ago
ThatTotallyRealMyth	05aa8c4e25	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	6 months ago
George Joseph	709be42bc3	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.	6 months ago
Alexei Gradinari	dfe25fbc8a	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	6 months ago
Sven Kube	280b13a053	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.	6 months ago
Naveen Albert	1e0205fb68	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	6 months ago
Tinet-mucw	47072a6f03	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	7 months ago
Sperl Viktor	5441b01193	res_agi: Increase AGI command buffer size from 2K to 8K ... Fixes: #1317	7 months ago
Naveen Albert	393e51495c	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	7 months ago
George Joseph	c8cacfba5e	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	7 months ago
George Joseph	f6d2c781a4	.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.	7 months ago
George Joseph	341c9b35e0	.github: Add skip-cherry-pick and skip-test-builds to Releaser.	7 months ago
George Joseph	a4da66e584	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	7 months ago
Naveen Albert	f5d2ea75c3	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	7 months ago
kodokaii	f03421b03b	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	7 months ago
George Joseph	08e19d0815	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	7 months ago
Martin Tomec	b333ee3be7	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	7 months ago
George Joseph	733196abf9	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.	7 months ago
George Joseph	93034831da	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	7 months ago
Stanislav Abramenkov	63e5b117ff	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	7 months ago
Michal Hajek	355ce73e72	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	7 months ago
Sean Bright	efd4be981b	channelstorage_makeopts.xml: Remove errant XML character. ... Resolves: #1282	7 months ago
mkmer	6d06d1765f	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	7 months ago
Tinet-mucw	bf9a6d80f8	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	7 months ago
Sean Bright	65001d26a6	res_musiconhold.c: Ensure we're always locked around music state access.	7 months ago
Sean Bright	a49d96505b	res_musiconhold.c: Annotate when the channel is locked.	7 months ago
Jaco Kroon	7fa200bba0	res_musiconhold: Appropriately lock channel during start. ... This relates to #829 This doesn't sully solve the Ops issue, but it solves the specific crash there. Further PRs to follow. In the specific crash the generator was still under construction when moh was being stopped, which then proceeded to close the stream whilst it was still in use. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	7 months ago
George Joseph	4852a046e7	res_stir_shaken.so: Handle X5U certificate chains. ... The verification process will now load a full certificate chain retrieved via the X5U URL instead of loading only the end user cert. * Renamed crypto_load_cert_from_file() and crypto_load_cert_from_memory() to crypto_load_cert_chain_from_file() and crypto_load_cert_chain_from_memory() respectively. * The two load functions now continue to load certs from the file or memory PEMs and store them in a separate stack of untrusted certs specific to the current verification context. * crypto_is_cert_trusted() now uses the stack of untrusted certs that were extracted from the PEM in addition to any untrusted certs that were passed in from the configuration (and any CA certs passed in from the config of course). Resolves: #1272 UserNote: The STIR/SHAKEN verification process will now load a full certificate chain retrieved via the X5U URL instead of loading only the end user cert.	8 months ago
George Joseph	2346803cce	res_stir_shaken: Add "ignore_sip_date_header" config option. ... UserNote: A new STIR/SHAKEN verification option "ignore_sip_date_header" has been added that when set to true, will cause the verification process to not consider a missing or invalid SIP "Date" header to be a failure. This will make the IAT the sole "truth" for Date in the verification process. The option can be set in the "verification" and "profile" sections of stir_shaken.conf. Also fixed a bug in the port match logic. Resolves: #1251 Resolves: #1271	8 months ago
Naveen Albert	8ee49a67f5	app_record: Add RECORDING_INFO function. ... Add a function that can be used to retrieve info about a previous recording, such as its duration. This is being added as a function to avoid possibly trampling on dialplan variables, and could be extended to provide other information in the future. Resolves: #548 UserNote: The RECORDING_INFO function can now be used to retrieve the duration of a recording.	8 months ago
Itzanh	d3a9bd29da	app_sms.c: Fix sending and receiving SMS messages in protocol 2 ... This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2. - Fix MORX message reception (from phone to Asterisk) in SMS protocol 2 - Fix MTTX message transmission (from Asterisk to phone) in SMS protocol 2 One of the bugs caused messages to have random characters and junk appended at the end up to the character limit. Another bug prevented Asterisk from sending messages from Asterisk to the phone at all. A final bug caused the transmission from Asterisk to the phone to take a long time because app_sms.c did not hang up after correctly sending the message, causing the phone to have to time out and hang up in order to complete the message transmission. This was tested with a Linksys PAP2T and with a GrandStream HT814, sending and receiving messages with Telefónica DOMO Mensajes phones from Telefónica Spain. I had to play with both the network jitter buffer and the dB gain to get it to work. One of my phones required the gain to be set to +3dB for it to work, while another required it to be set to +6dB. Only MORX and MTTX were tested, I did not test sending and receiving messages to a TelCo SMSC.	8 months ago
phoneben	a31b68c37a	app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range ... This update adds support for a new QUEUE_RAISE_PENALTY format: rN When QUEUE_RAISE_PENALTY is set to rN (e.g., r4), only members whose current penalty is greater than or equal to the defined min_penalty and less than or equal to max_penalty will have their penalty raised to N. Members with penalties outside the min/max range remain unchanged. Example behaviors: QUEUE_RAISE_PENALTY=4 → Raise all members with penalty < 4 (existing behavior) QUEUE_RAISE_PENALTY=r4 → Raise only members with penalty in [min_penalty, max_penalty] to 4 Implementation details: Adds parsing logic to detect the r prefix and sets the raise_respect_min flag Modifies the raise logic to skip members outside the defined penalty range when the flag is active UserNote: This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises only for members whose current penalty is within the [min_penalty, max_penalty] range. Members with lower or higher penalties are unaffected. This behavior is backward-compatible with existing queue rule configurations.	8 months ago
George Joseph	75e2465069	res_websocket_client: Add more info to the XML documentation. ... Added "see-also" links to chan_websocket and ARI Outbound WebSocket and added an example configuration for each.	8 months ago
Jaco Kroon	c8ab8be1ca	res_odbc: cache_size option to limit the cached connections. ... Signed-off-by: Jaco Kroon <jaco@uls.co.za> UserNote: New cache_size option for res_odbc to on a per class basis limit the number of cached connections. Please reference the sample configuration for details.	8 months ago
Jaco Kroon	c684603d72	res_odbc: cache_type option for res_odbc. ... This enables setting cache_type classes to a round-robin queueing system rather than the historic stack mechanism. This should result in lower risk of connection drops due to shorter idle times (the first connection to go onto the stack could in theory never be used again, ever, but sit there consuming resources, there could be multiple of these). And with a queue rather than a stack, dead connections are guaranteed to be detected and purged eventually. This should end up better balancing connection_cnt with actual load over time, assuming the database doesn't keep connections open excessively long from it's side. Signed-off-by: Jaco Kroon <jaco@uls.co.za> UserNote: When using res_odbc it should be noted that back-end connections to the underlying database can now be configured to re-use the cached connections in a round-robin manner rather than repeatedly re-using the same connection. This helps to keep connections alive, and to purge dead connections from the system, thus more dynamically adjusting to actual load. The downside is that one could keep too many connections active for a longer time resulting in resource also begin consumed on the database side.	8 months ago
Sean Bright	7de37418fd	res_pjsip: Fix empty `ActiveChannels` property in AMI responses. ... The logic appears to have been reversed since it was introduced in 05cbf8df. Resolves: #1254	8 months ago
George Joseph	65199303b9	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	8 months ago
George Joseph	831c961a63	res_websocket_client: Create common utilities for websocket clients. ... Since multiple Asterisk capabilities now need to create websocket clients it makes sense to create a common set of utilities rather than making each of those capabilities implement their own. * A new configuration file "websocket_client.conf" is used to store common client parameters in named configuration sections. * APIs are provided to list and retrieve ast_websocket_client objects created from the named configurations. * An API is provided that accepts an ast_websocket_client object, connects to the remote server with retries and returns an ast_websocket object. TLS is supported as is basic authentication. * An observer can be registered to receive notification of loaded or reloaded client objects. * An API is provided to compare an existing client object to one just reloaded and return the fields that were changed. The caller can then decide what action to take based on which fields changed. Also as part of thie commit, several sorcery convenience macros were created to make registering common object fields easier. UserNote: A new module "res_websocket_client" and config file "websocket_client.conf" have been added to support several upcoming new capabilities that need common websocket client configuration.	8 months ago
George Joseph	4e83f99f76	asterisk.c: Add option to restrict shell access from remote consoles. ... UserNote: A new asterisk.conf option 'disable_remote_console_shell' has been added that, when set, will prevent remote consoles from executing shell commands using the '!' prefix. Resolves: #GHSA-c7p6-7mvq-8jq2	9 months ago
George Joseph	72cff16c76	res_pjsip_messaging.c: Mask control characters in received From display name ... Incoming SIP MESSAGEs will now have their From header's display name sanitized by replacing any characters < 32 (space) with a space. Resolves: #GHSA-2grh-7mhv-fcfw	9 months ago
mkmer	eb427c6e4a	frame.c: validate frame data length is less than samples when adjusting volume ... Resolves: #1230	9 months ago