asterisk

Commit Graph

Author	SHA1	Message	Date
Sperl Viktor	b88e04a6f5	res_agi: Increase AGI command buffer size from 2K to 8K Fixes: #1317 (cherry picked from commit `3403da9c01`)	3 weeks ago
Naveen Albert	6646bc2dfd	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 (cherry picked from commit `9c55e72b67`)	3 weeks ago
George Joseph	10d4f2c45f	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 (cherry picked from commit `a4aab9d39f`)	3 weeks ago
George Joseph	263a9f6ee0	.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. (cherry picked from commit `f459a57394`)	3 weeks ago
George Joseph	10b611d287	.github: Add skip-cherry-pick and skip-test-builds to Releaser. (cherry picked from commit `cc74f2f28c`)	3 weeks ago
George Joseph	33885d8893	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 (cherry picked from commit `8a602dbfee`)	3 weeks ago
Naveen Albert	a9f67ab6c9	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 (cherry picked from commit `a45e6d6fd8`)	3 weeks ago
kodokaii	e0552b1c8e	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 (cherry picked from commit `38d0909669`)	3 weeks ago
Martin Tomec	1d47c4c7db	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 (cherry picked from commit `5946bc6363`)	3 weeks ago
George Joseph	df33c52a1f	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. (cherry picked from commit `07fd3af897`)	3 weeks ago
Stanislav Abramenkov	48e058d0f1	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 (cherry picked from commit `877e5ef770`)	3 weeks ago
mkmer	9232cde023	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 (cherry picked from commit `55f4294e82`)	3 weeks ago
Tinet-mucw	3b9d5a00e4	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 (cherry picked from commit `88f653c12d`)	3 weeks ago
Sean Bright	386a84a76f	res_musiconhold.c: Ensure we're always locked around music state access. (cherry picked from commit `7feb676409`)	3 weeks ago
Sean Bright	d1d66dfe16	res_musiconhold.c: Annotate when the channel is locked. (cherry picked from commit `def223ffd8`)	3 weeks ago
Jaco Kroon	b758f2e880	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> (cherry picked from commit `9725811125`)	3 weeks ago
Asterisk Development Team	f34af39805	Update for 20.15.2	3 weeks ago
George Joseph	9d7048083b	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	3 weeks ago
Asterisk Development Team	21d22b328a	Update for 20.15.1	3 weeks ago
ThatTotallyRealMyth	0788c03f92	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	3 weeks ago
George Joseph	53b571ae6e	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	3 weeks ago
George Joseph	ac25398707	.lastclean: Remove from git It never should been versioned in the first place.	3 weeks ago
Asterisk Development Team	be0642f6ba	Update for 20.15.0	3 weeks ago
Asterisk Development Team	87b58f398b	Update for 20.15.0-rc3	3 weeks ago
George Joseph	ca7db896e5	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 weeks ago
George Joseph	abdfb5aa87	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 weeks ago
Asterisk Development Team	bd9e6e1e01	Update for 20.15.0-rc2	3 weeks ago
Michal Hajek	7f7c748762	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 weeks ago
Sean Bright	aeaeb26438	channelstorage_makeopts.xml: Remove errant XML character. Resolves: #1282	3 weeks ago
Asterisk Development Team	4665bdcee0	Update for 20.15.0-rc1	3 weeks ago
George Joseph	00b99494b4	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.	3 weeks ago
George Joseph	12337a55a4	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	3 weeks ago
Naveen Albert	eb6808e580	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.	3 weeks ago
Itzanh	2928b876d8	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.	3 weeks ago
phoneben	76e22b0670	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.	3 weeks ago
George Joseph	63656c1f8f	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.	3 weeks ago
Jaco Kroon	3ef106d3a9	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.	3 weeks ago
Jaco Kroon	6dbeace0c7	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.	3 weeks ago
Sean Bright	5ca12d6cdc	res_pjsip: Fix empty `ActiveChannels` property in AMI responses. The logic appears to have been reversed since it was introduced in `05cbf8df`. Resolves: #1254	3 weeks ago
George Joseph	f82c7b2694	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	3 weeks ago
George Joseph	04a3e854d0	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.	3 weeks ago
George Joseph	46549fa4e3	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	3 weeks ago
George Joseph	07cee24a75	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	3 weeks ago
mkmer	642504c007	frame.c: validate frame data length is less than samples when adjusting volume Resolves: #1230	3 weeks ago
Sven Kube	3924cdc975	res_audiosocket.c: Add retry mechanism for reading data from AudioSocket The added retry mechanism addresses an issue that arises when fragmented TCP packets are received, each containing only a portion of an AudioSocket packet. This situation can occur if the external service sending the AudioSocket data has Nagle's algorithm enabled.	3 weeks ago
Sven Kube	2d404f461c	res_audiosocket.c: Set the TCP_NODELAY socket option Disable Nagle's algorithm by setting the TCP_NODELAY socket option. This reduces latency by preventing delays caused by packet buffering.	3 weeks ago
Thomas B. Clark	c4aad078fb	menuselect: Fix GTK menu callbacks for Fedora 42 compatibility This patch resolves a build failure in `menuselect_gtk.c` when running `make menuconfig` on Fedora 42. The new version of GTK introduced stricter type checking for callback signatures. Changes include: - Add wrapper functions to match the expected `void (*)(void)` signature. - Update `menu_items` array to use these wrappers. Fixes: #1243	3 weeks ago
Stanislav Abramenkov	ac1fd7ffc1	jansson: Upgrade version to jansson 2.14.1 UpgradeNote: jansson has been upgraded to 2.14.1. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.14.1 Resolves: #1178	3 weeks ago
Joe Searle	8881655fa2	pjproject: Increase maximum SDP formats and attribute limits Since Chrome 136, using Windows, when initiating a video call the INVITE SDP exceeds the maximum number of allowed attributes, resulting in the INVITE being rejected. This increases the attribute limit and the number of formats allowed when using bundled pjproject. Fixes: #1240	3 weeks ago
Nathan Monfils	84a7cae3b1	manager.c: Invalid ref-counting when purging events We have a use-case where we generate a lot of events on the AMI, and then when doing `manager show eventq` we would see some events which would linger for hours or days in there. Obviously something was leaking. Testing allowed us to track down this logic bug in the ref-counting on the event purge. Reproducing the bug was not super trivial, we managed to do it in a production-like load testing environment with multiple AMI consumers. The race condition itself: 1. something allocates and links `session` 2. `purge_sessions` iterates over that `session` (takes ref) 3. `purge_session` correctly de-referencess that session 4. `purge_session` re-evaluates the while() loop, taking a reference 5. `purge_session` exits (`n_max > 0` is false) 6. whatever allocated the `session` deallocates it, but a reference is now lost since we exited the `while` loop before de-referencing. 7. since the destructor is never called, the session->last_ev->usecount is never decremented, leading to events lingering in the queue The impact of this bug does not seem major. The events are small and do not seem, from our testing, to be causing meaningful additional CPU usage. Mainly we wanted to fix this issue because we are internally adding prometheus metrics to the eventq and those leaked events were causing the metrics to show garbage data.	3 weeks ago

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