asterisk

Commit Graph

Author	SHA1	Message	Date
Stanislav Abramenkov	ee58f361c7	jansson: Upgrade version to jansson 2.15.1 UpgradeNote: jansson has been upgraded to 2.15.1. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.1 Resolves: #2033	2 months ago
Jeremy Lainé	1d164595d4	file.c: Ensure opening a stream opens at most one file The functions used to stream files eventually end up calling into `filehelper(.., ACTION_OPEN)` which takes care of iterating over supported extensions and opening the first existing file. To do this, `filehelper` uses two nested loops: - An outer loop over the supported file formats. - An inner loop over the possible extensions for that format. We need to break out of both loops as soon as a file was successfully opened. Otherwise if a file exists in multiple formats (e.g `foo.wav` and `foo.alaw`) both files will be opened successively. Resolves: #1625	2 months ago
Joshua C. Colp	9f24612ba8	pjsip: Eliminate some unique taskprocessors. When placing outbound calls each call would get its own taskprocessor for things related to the call. In practice this is overkill as few things actually occur. Inbound calls on the other hand already use one of the fixed number of distributor taskprocessors. This also occurred for OPTIONS requests with each AOR having its own taskprocessor. This change moves both to using a distributor taskprocessor instead.	2 months ago
Roel van Meer	2b85c53cc5	bridging: Add support for TOUCH_MIXMONITOR_OPTIONS This channel variable can be used to override the default automixmon options (which are 'b' or 'bB(<n>)'). If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning. It is the responsibility of the user to configure valid options in TOUCH_MIXMONITOR_OPTIONS. Fixes: #1981 UserNote: The TOUCH_MIXMONITOR_OPTIONS channel variable can now be used to configure the options used by the mixmonitor application when it is started via automixmon. If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning.	2 months ago
George Joseph	da3de0d905	res_pjsip_refer: Fix issues with Refer-Sub:false and ARI. When an attended transfer is controlled by ARI using PJSIP_TRANSFER_HANDLING()=ari-only, and a REFER request is received with a Refer-Sub: false header to suppress the automatic progress subscription, a segfault occurs in res_pjsip_refer:refer_incoming_ari_request(). However, even if the segfault were prevented, there's another issue... Suppressing the subscription and NOTIFYs to the referer also prevents the ARI app from getting any progress events for the transfer which it's controlling. So... * The segfault has been fixed. * Suppressing the subscription no longer suppress the ARI events from being sent to the controlling ARI app. * A good amount of tracing has also been added. Resolves: #2021	2 months ago
Mike Bradeen	ab63dece36	build: re-add pjsua test application pjsua and it's python bindings were removed as part of PR 1854. Testing for RTT requires the pjsua application be re-added to the third pary build process. This applies to the pjsua application ONLY and not the associated python bindings. Resolves: #2024	2 months ago
Naveen Albert	80e3ac64e5	res_pjsip: Document 'none' option for 'dtmf_mode'. The 'none' value can be used to disable DSP processing for DTMF on PJSIP channels and is sometimes necessary for this reason, and already exists in the code, but is not documented. Add it to the documentation enum. Resolves: #2026	2 months ago
Mike Bradeen	6916dfe360	Audiohooks: Allow whisper audiohooks to work without an underlying stream Audiohook whispers work by mixing audio with the current stream of frames being written to the hooked channel. Prior to this change if there were no audio frames being written to the channel outside of the audiohook, then any whisper audio has nothing to mix with and so is not heard on the hooked channel. This change removes that requirement by creating a framehook driven timer on the channel that creates silent frames the whispered frames can be mixed into. The framehook is automatically added with the first attached whisper audiohook and is removed when the last one is detached. If there are no attached streams being written to the channel, the timer is used to generate a default one. If a stream is attached; the timer is ignored until the stream stops. This stop and start is automatically triggered by the presence of a written stream and does not require any external action. Resolves: #1966 UserNote: There is no longer a requirement to play silence or otherwise have a stream of audio being sent to a hooked channel for whispered frames to be heard.	2 months ago
Ben Ford	394b324871	Logger: Don't discard WARNING or ERROR messages. When the logger reaches it's threshold for messages, don't discard the WARNING or ERROR messages up to a certain amount. If THAT threshold reaches its maximum, start discarding the oldest non-WARNING/ERROR messages until we only have WARNING and ERROR messages left. Everything will be discarded after that until we have room again for more. Also added a test that limits the queue sizes and logs many messages to see the queues hitting their maximums and discarding the appropriate messages. Fixes: #1882	3 months ago
Aqeel Abbas	2770e2bb83	func_strings: Fix syntax error in STRBETWEEN documentation example The STRBETWEEN function documentation example was missing a closing brace in the SendDTMF call. Corrected the syntax to ensure the dialplan example is functional.	3 months ago
Mehrdad Seifzadeh	a6fdedebae	res_pjsip_registrar: Resolve AOR for identified endpoints When an inbound REGISTER is matched to an endpoint using an identification method such as IP, header, or request URI, the endpoint can be identified and authenticated successfully while registrar AOR resolution still fails. find_registrar_aor() only resolved AOR names for username and auth_username identify methods. A header-only endpoint therefore left aor_name unset and failed registration with "AOR '' not found" even when the REGISTER To URI matched one of the endpoint's configured AORs. Resolve the REGISTER AOR from the To URI for endpoint identification methods that do not directly provide an AOR name, while still constraining the match to the already-identified endpoint's configured AOR list. Fixes: #1398	3 months ago
Thomas Guebels	0330d37340	res_http_websocket: Unref session when it fails to establish. Two error paths in res_http_websocket could leak an ast_websocket session by failing to drop a reference after the session had been allocated. The first occurs when session ID generation fails. While this is largely theoretical, as it would require an out-of-memory condition, the session reference should still be released correctly. The second occurs when the ast_websocket_pre_callback session_attempted callback returns an error. This path is reachable through ari_websockets, which implements this callback and can legitimately fail under certain conditions. Fixes: #2020	3 months ago
George Joseph	b4cd1ec359	Bundled pjproject: Make it easier to override options in config_site.h. Bundled pjproject uses ./third-party/pjproject/patches/config_site.h to set many pjproject build options like PJ_IOQUEUE_MAX_HANDLES, PJSIP_MAX_URL_SIZE, etc. Editing that file however, causes the git tree to become dirty which is inconvenient. So... * Updated the Asterisk configure scripts to pass the PJPROJECT_CFLAGS variable down to the pjproject configure scripts. See the UserNote below for details. * Updated ./third-party/pjproject/patches/config_site.h to allow the following options to be overridden: PJ_MAX_HOSTNAME, PJSIP_MAX_URL_SIZE, PJ_IOQUEUE_MAX_HANDLES. Other options in config_site.h are not overridable because they can have a bad effect on the overall operation of pjproject. This may be revisited in the future. Options not already set in config_site.h can still be set. * Fixed an issue where if the Linux `epoll` facility is used (which it is by default) the default PJ_IOQUEUE_MAX_HANDLES and PJSIP_MAX_TRANSPORTS were being left at 1024 instead of being icnreased to 5000. * The `pjproject show buildopts` CLI command previously only showed options from the top level `PJ` pjproject layer but now also shows many from the `PJSIP` layer. Many of these, such as PJSIP_MAX_TRANSPORTS, can be set using PJPROJECT_CFLAGS provided they're not already unconditionally set in config_site.h. * The `pjsip dump endpt` CLI command previously required that the pjproject log level be already set to at least 3 or no output would be produced. The command now does that automatically then sets it back to whatever it was. This isn't strictly related to this PR but was just nagging me. UserNote: Bundled pjproject: It's now possible to override some of the pjproject build options contained in ./third-party/pjproject/patches/config_site.h by adding PJPROJECT_CFLAGS to your Asterisk ./configure command line. For example: `./configure ... PJPROJECT_CFLAGS='-DPJ_OPT1=8192 -DPJ_OPT2=512'` Any option in config_site.h that's wrapped in a `#ifndef` block can be overridden and many of the `PJSIP` options displayed by `pjproject show buildopts` can be set. WARNING: Adjusting these options without understanding their effect can cripple your Asterisk instances. You shouldn't adjust them unless you need to solve a specific issue. Resolves: #2011	3 months ago
Joshua C. Colp	2363297479	test_performance: Add performance experimentation test module. This is a module which is only built when TEST_FRAMEWORK is enabled and provides CLI commands for testing performance of certain things on the system they are invoked on. The first CLI commands added cover the most common container usage: storage of objects with a lookup based on a string key. These commands take various arguments and allow you to see how they perform. There is also an "all" command named "performance test container_key_lookup_all" that will execute all of these container tests and pass through the given arguments, which makes it easy to run all of the tests for given usage. To facilitate a vector bsearch test a new AST_VECTOR_BSEARCH macro has been added that allows more efficient searching of sorted vectors.	3 months ago
Joshua C. Colp	0a774c6223	manager: Move away from shared linked list for events. This change moves manager away from a shared linked list for events to a per-session vector of pending events. This allows early filtering of events which eliminates the need to wake up threads for events that they have no interest in. TCP based connections have also been moved to an alert pipe based wakeup mechanism to reduce locking and simplify usage. UserNote: The "manager show eventq" CLI command has been removed as there is no longer a single event queue to display.	3 months ago
George Joseph	26551a4dc0	chan_websocket: Use leftover data if no frames are available when the timer fires. When the 20ms channel timer fires but there are no frames available in the queue, we now check for leftover data in the buffer and if there is any, we create a frame with it and send it to the core. This resolves an issue with the leftover data being delayed if a STOP_MEDIA_BUFFERING command is delayed. Some existing comments were also clarified to account for the new behavior. Resolves: #2006	3 months ago
Mehrdad Seifzadeh	05dc5b2d6f	res_pjsip_session: Bound delayed BYE behind UAC INVITE When a confirmed session is being terminated while an outgoing in-dialog INVITE transaction is still outstanding, the BYE is delayed until the outstanding transaction terminates. If that INVITE has already received a provisional response and the final response is malformed and rejected before transaction processing, the transaction can remain outstanding and the delayed BYE can keep the session, media state, RTP instance, and PJPROJECT pools referenced after the channels are gone. When a BYE is delayed behind an outstanding UAC INVITE, set a PJPROJECT transaction timeout on that INVITE so the delayed cleanup path has a bounded wait. If PJPROJECT terminates the dialog as a result of the timeout, discard the delayed BYE instead of sending a duplicate BYE. Fixes: #1965	3 months ago
Alexandre Fournier	6b703b12d5	format_cap: guard against NULL src in _from_cap helpers ast_format_cap_append_from_cap() and ast_format_cap_replace_from_cap() dereference 'src' (src->preference_order) without checking it for NULL. A dummy channel allocated with ast_dummy_channel_alloc() never sets a native-format capability, so ast_channel_nativeformats() returns NULL on such channels. When CHANNEL(audionativeformat) / CHANNEL(videonativeformat) is evaluated against a dummy channel (e.g. via ARI channelvars during a Stasis VarSet event raised while app_voicemail builds the notification email on a dummy channel), func_channel_read() passes that NULL straight into ast_format_cap_append_from_cap(), causing a NULL dereference at offset 0x28 and a SIGSEGV. Guard both helpers against a NULL source. A NULL source simply means "no formats to copy", so appending/replacing nothing is the correct no-op behaviour. This also protects all other callers. Fixes https://github.com/asterisk/asterisk/issues/1992 AI disclosure: this was generated using Claude Opus 4.8, tested to fix the issue. Not sure if it is the right* way to do it.	3 months ago
ThatTotallyRealMyth	fe4d430ada	ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory. The ast_tsconvert.py script called by ast_loggrabber is now installed in a temporary directory that isn't world readable or writable. Resolves: #GHSA-xgj6-2gc5-5x9c	3 months ago
George Joseph	136b3adfcd	chan_unistim.c: Prevent overrun of phone_number field. Add a check to key_dial_page() to ensure that dialed digits won't overrun the phone_number field. Resolves: #GHSA-3g56-cgrh-95p5	3 months ago
George Joseph	b66aeb6c0c	res_ari: Ensure read-only users are properly authorized via REST Over WebSocket. The REST over WebSocket path now properly prevents non-GET methods from being executed on inbound WebSockets. * The query parameters from the original incoming GET request that caused the upgrade to WebSocket are now passed to all REST requests that come from the client. This ensures that if the client authenticated with a read-only userid using the "api_key" query_string parameter, REST requests coming in over the WebSocket will only be able to execute GETs on resources. The HTTP headers were already passed to the REST requests so if the client had authenticated via an "Authorization" it was properly handled. * New tests have been added to test_ari.c to check that read-only users are properly denied access to resources using non-GET methods. Several memory leaks were also squashed. Resolves: #GHSA-wcvv-g26m-wx5c	3 months ago
George Joseph	79b4dc837e	pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address. The filter_on_tx_message() function was using pj_strassign() to save the pointer of the pjproject transport local address to a local pj_str_t variable. That variable was ultimately used to set the Contact header's uri->host and the SDP connection attribute's address again using pj_strassign. pj_strassign() doesn't copy the actual value of the pj_str_t however, it just copies the pointer so if a connection-oriented transport is disconnected before the 200 OK with the SDP is sent, those pointers will be invalid which can cause use-after-free issues. To prevent this, filter_on_tx_message() now uses pj_strdup with the tdata->pool as the backing store to save the local IP address to the local variable. pj_strassign() can then be used safely later on since the tdata will be available for the life of the transaction. Resolves: #GHSA-g8q2-p36q-94f6	3 months ago
George Joseph	373b4e21de	ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds. Several bounds checks have been edded to ooQ931Decode to prevent it from running past the end of the data buffer when parsing information elements. Resolves: #GHSA-746q-794h-cc7f	3 months ago
George Joseph	437eb40e84	ARI: Make ARI applications respect live_dangerously. DeveloperNote: ARI applications can no longer call "dangerous" dialplan functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling "live_dangerously" in asterisk.conf. Resolves: #GHSA-vrfp-mg3q-3959	3 months ago
George Joseph	ce61518203	res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns. * Add check to red_t140_to_red() to ensure that the new primary payload can't cause the rtp_red->len array items to wrap or cause an overrun of the rtp_red->t140red_data buffer. * Add check to rtp_red_buffer() to ensure that a T.140 frame to be sent can't cause rtp_red->len array items to wrap or cause an overrun of the rtp_red->buf_data buffer. Resolves: #GHSA-vfhr-r9x9-c687 Resolves: #GHSA-j2mm-57pq-jh94	3 months ago
Roberto Paleari	ce42d5d3f2	res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser Add constraint checks to prevent unauthenticated users from crashing Asterisk instance by sending a crafted inbound SIP NOTIFY request with "Content-Type: application/simple-message-summary". Resolves: #GHSA-8jw3-ccr9-xrmf	3 months ago
Mike Bradeen	fbaabcaaa2	manager: Use remote address in user error logging To avoid a potential null dereference use the remote address in error logging when there is no user or the user acl fails. Resolves: #GHSA-3rhj-hhw7-m6fw	3 months ago
Mike Bradeen	70b0abcd03	ooh323: Prevent potential buffer overflow in trace logging Replace a call to vsprintf with a call to ast_vasprintf to prevent a possible buffer overflow. Resolves: #GHSA-x348-j6c9-77f3	3 months ago
Pengpeng Hou	9724288770	app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers The protocol 1 unpack helpers trusted externally controlled lengths and wrote them directly into fixed-size buffers in sms_t. Clamp the address, header, and body copies to the destination array sizes so malformed messages cannot overwrite adjacent state. Resolves: #GHSA-q9fr-m7g8-6ph5	3 months ago
Milan Kyselica	8bcf6b79f8	res_xmpp: Fix stack buffer overflow in namespace prefix handling The snprintf size parameter in xmpp_action_hook() is computed from the attacker-controlled namespace prefix length and is not bounded by the 256-byte stack buffer size. When a remote XMPP peer sends a stanza with a child element whose namespace prefix exceeds 249 characters, snprintf writes past the buffer boundary. Use sizeof(attr) as the snprintf size limit and %.*s precision to extract only the prefix portion of the element name, preserving the original truncation behavior for valid inputs. Resolves: #GHSA-mxgm-8c6f-5p8f	3 months ago
Milan Kyselica	2121ff8ef6	res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser The parse_simple_message_summary() function uses sscanf with an unbounded %s format specifier to parse the Message-Account field from incoming SIP NOTIFY bodies into a fixed-size 512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single unauthenticated SIP NOTIFY with a Message-Account value exceeding 512 bytes overflows the buffer, corrupting adjacent stack data and permanently disabling the PJSIP transport layer without crashing the process. Add a width specifier (%511s) to limit the sscanf write to PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator, matching the destination buffer size. Resolves: #GHSA-589g-qgf8-m6mx	3 months ago
Milan Kyselica	56deaaa063	res_config_ldap: Escape LDAP filter values per RFC 4515 The LDAP realtime driver constructs search filters by directly concatenating user-supplied values without RFC 4515 escaping. When LDAP is used as a realtime backend for endpoint identification, characters with special meaning in LDAP filters (*, (, ), \) can be injected via the SIP From header username. Add ldap_filter_escape_value() that escapes RFC 4515 special characters to their \HH hex representation, and apply it to non-LIKE query values. The LIKE query path preserves the existing wildcard conversion behavior with a note for maintainers. Resolves: #GHSA-r6c2-hwc2-j4mp	3 months ago
Milan Kyselica	8f51e97808	cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection The eventtype column handler in cel_pgsql.c inserts record.user_defined_name directly into the SQL query without calling PQescapeStringConn(), while all other string fields in the same function are properly escaped. Similarly, cel_tds.c passes the raw user_defined_name into the SQL INSERT without routing it through anti_injection(), while all other fields are processed through that function. For cel_pgsql.c, escape the eventtype value using PQescapeStringConn(), matching the existing pattern used for all other string fields at lines 308-331 of the same function. For cel_tds.c, route the eventtype value through anti_injection() consistent with how all other fields are handled in the same function. Resolves: #GHSA-ph27-3m5q-mj5m	3 months ago
Milan Kyselica	98f50c235e	http: Escape error page text to prevent reflected XSS The text parameter in ast_http_create_response() is inserted into the HTML body without escaping, while the server name on the same page is properly escaped via ast_xml_escape(). When res_phoneprov passes the decoded request URI as the text of a 404 response, HTML metacharacters in the URI are rendered by the browser. Apply ast_xml_escape() to the text parameter before inserting it into the HTML template, using the same function already used for the server name. Resolves: #GHSA-4pgv-j3mr-3rcp	3 months ago
Milan Kyselica	7a8d4975fe	codec_codec2: Only process complete Codec2 frames in decoder The codec2_samples() function uses floor division (160 * datalen/6) to compute expected output samples, but the decode loop condition (x < datalen) iterates with ceiling behavior when datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch causes the loop to decode one extra frame beyond what the framework bounds check budgeted for, leading to an out-of-bounds write on the output buffer. Change the loop condition to only process complete frames, matching the floor-division behavior of codec2_samples(). This also prevents an out-of-bounds read on the input side when fewer than CODEC2_FRAME_LEN bytes remain. Resolves: #GHSA-qf8j-jp7h-c5hx	3 months ago
Milan Kyselica	2f38860a5a	format_ogg_speex: Add bounds check to prevent heap buffer overflow The ogg_speex_read() function copies OGG packet data via memcpy() without validating the packet size against the destination buffer (BUF_SIZE = 200 bytes). A crafted .spx file with an oversized OGG audio packet causes a heap buffer overflow that corrupts the adjacent speex_desc structure containing libogg heap pointers, leading to a crash (SIGSEGV) on playback. Add a bounds check for both negative and oversized values before the memcpy, consistent with how format_ogg_vorbis bounds its reads via ov_read(). Resolves: #GHSA-8jhw-m2hg-vp3h	3 months ago
Sean Bright	28e7566f88	configs: Comment out `values` setting to avoid parse error Fixes the following after a `make samples`: ``` config.c:2281 process_text_line: parse error: No category context for line 64 of ... ```	3 months ago
Mike Bradeen	3bd8695703	app_mixmonitor: Fix duplex recording for non 8K codecs The native sampling of duplex recording is set to match the raw 8K output format. If one or more of the streams being recorded is above 8K, the frame size coming into the mixmonitor is too large and needs to be translated to 8K before being mixed into the stereo frame to avoid garbled and mistimed audio Fixes: #1779	3 months ago
George Joseph	9173b0f526	res_http_websocket: Add timeout to client handshakes. The websocket client proxy and server handshakes use ast_iostream_gets which are blocking calls. If the outgoing connection succeeds at the TCP or TLS layer but the proxy (if configured) or the websocket server fails to respond to the CONNECT or GET requests, the process can hang indefinitely and escalate to a deadlock. To address this, the handshakes are now guarded with calls to ast_iostream_set_timeout_sequence() with the timeout set to the client's (connection_timeout * 2) milliseconds. In order to use ast_iostream_set_timeout_sequence(), the iostream has to be set to non-blocking with ast_iostream_nonblock() but there was no way to reset the stream back to blocking mode so a new API ast_iostream_blocking() was added for it. Tracing was also enabled in the websocket_client_handshake function for future troubleshooting. Resolves: #1979	3 months ago
Joshua C. Colp	d40369adc3	extension_state: Add new extension state API. Extension state to this point has been an API implemented inside the PBX core resulting in its state being intermingled with that of the dialplan. This increased the complexity of the PBX core and made it difficult to enact improvements. This change adds a new separate extension state API which receives updates from the PBX core as configuration changes but maintains its own separate state. The API is also written to fully take advantage of modern APIs in a more selective manner by subscribing each extension state to only the devices it is interested in, ultimately reducing resource consumption during updates. Presence state updates being infrequently done use a single shared subscription that goes through the extension states to find and update ones that the update is applicable to. Legacy API support is provided by reimplementing the API as wrappers over the new extension state API. This improves the legacy API by making it multithreaded, with each callback being individually subscribed. Autohints support is maintained but has been separated out into a self contained new implementation. Synchronous subscription support has also been added to Stasis to remove the overhead of asynchronous publishing when the handling of published messages is small and fast.	4 months ago
Alexis Chenard	052409f4b6	res_pjsip: Add external_signaling_hostname transport option Adds a new transport option 'external_signaling_hostname' which allows a hostname or FQDN to be used in SIP Contact and Via headers instead of the automatically determined IP address. This is useful when a remote SIP endpoint requires a fully qualified domain name in these headers. The option is mutually exclusive with 'external_signaling_address' and an error is raised at transport load time if both are set simultaneously. Resolves: #1749 UserNote: A new pjsip.conf transport option 'external_signaling_hostname' has been added. When set, this value will be used in SIP Contact and Via headers instead of the automatically determined IP address. This option is mutually exclusive with 'external_signaling_address'.	4 months ago
George Joseph	d8df207318	WebSocket Enhancements: Proxies and Keepalives for ARI and Media Outbound Websockets. See the notes below for high-level descriptions of the new features. * Proxies Outbound/forward HTTP proxies are now supported and configurable in websocket_client.conf. You can specify a host:port plus optional proxy_username and proxy_password. Because WebSockets aren't consistently supported among proxies (specifically passing through UPGRADEs), the CONNECT method is always used to establish a TCP tunnel through the proxy. This is required if a TLS session is to be established with the WebSocket server anyway. It's important to understand that that negotiation with the proxy is ALWAYS unsecured. Once the proxy establishes the tunnel, the TLS session will be negotiated directly with the remote WebSocket server via the tunnel. * Keepalives Both TCP-level and WebSocket PING/PONG keepalives can be configured and are available with either the curl or tcptls client implementations. The TCP keepalives are handled entirely by the operating system and require no resources from Asterisk but by their very nature, they can't traverse proxies. WebSocket PING/PONGs are implemented in the Asterisk websocket code and require a scheduler thread to keep track of them so they're a bit more complicated but they do traverse proxies. Which one is used is completely up to the admin. You could use both. * Other Changes A few changes were needed to res/ari/ari_websockets and res/res_aeap/transport_websocket to add explicit calls to ast_websocket_close. They had been assuming that the websocket session destructor would close the websocket when it unreffed it but the keepalive process now holds a reference so the destructor wouldn't actually run without the call to ast_websocket_close to stop the keepalives. A few new methods were added to tcptls.c to allow switching an existing connection from unsecured to TLS. These were required because the initial connection and handshake with a proxy is always unsecured but then needs to be switched to TLS if required for the remote WebSocket server. There was a bug in sorcery.h where the ast_sorcery_register_uint macro was referencing _stringify (which doesn't exist) instead of _sorcery_stringify. Resolves: #1881 Resolves: #1933 UserNote: Forward/outbound proxies can now be specified for outbound websockets. See the websocket_client.conf.sample file for configuration information. UserNote: TCP-level or WebSocket PING/PONG keepalives can now be enabled on outbound websockets. They can help detect network failures even when a persistent connection is idle. See the websocket_client.conf.sample file for configuration information. DeveloperNote: The addition of the proxy and keepalive configuration parameters pushed the websocket client parameter count over 32. This necessitated changing the size of the ast_ws_client_fields enum from a 32 bit bitfield to a 64-bit bitfield with a corresponding change to the ast_websocket_client structure.	4 months ago
Naveen Albert	922e31b107	chan_local: Update chan_local references for Local channels. chan_local no longer exists since Local channels are built into the core (core_local), but there are still comments which reference it, including in the configs. Update these to avoid confusion. Resolves: #1849	4 months ago
George Joseph	e3c0b816ea	res_ari: Add res_ari_model as an optional_module. Under certain timing/load conditions, res_ari_model may not load until after res_ari on startup or it might unload before res_ari on shutdown. This can cause a segfault when DEVMODE is enabled and there are persistent outbound websocket connections because DEVMODE forces validation of outgoing events against the models. To prevent this, res_ari_model has been added as an "optional_module" to res_ari's NODULE_INFO. This will enforce load/unload order but not make res_ari dependent on res_ari_model. However, if Asterisk is configured with --enable-dev-mode, res_ari will fail to load if res_ari_model isn't available. Resolves: #1970	4 months ago
Mike Bradeen	45198e9af0	res ari: Add attachable states to Channels and Bridges Adds the ability to attach multiple states to both Channels and Bridges in the form of variables that are included in all events on the associated object. First, this adds an optional boolean field to channel variables 'report_events' that causes the variable to automatically be included in all events on that channel. To allow this, variables can now be either name value pairs (the current format): `<variable_name>: '<value_string>'` - or - `<variable_name>: {value: '<value_string>', report_events: [true\|false]}` If the old format is used or 'report_events' is not included, it will default to false and retain current behavior. Second, this extends both reported and unreported variables to Bridges so they too may have stateful information. Resolves: #1910 UserNote: Bridge variables now can be set and retrieved via the following paths: `/bridges/{bridgeId}/variable` `/bridges/{bridgeId}/variables` Both Bridge and Channel variables can now be set with an optional 'report_events' boolean flag that will cause those variables to be included on all events on that object. The 'report_events' flag will default to False if not set to maintain backwards capability. To allow this, variables can now be either name value pairs (the current format): `<variable_name>: '<value_string>'` - or - `<variable_name>: {value: '<value_string>', report_events: [true\|false]}`	4 months ago
Ben Ford	6888844375	ARI: Added paths to get and set multiple channel variables. Two new paths exist for ARI to get and set multiple channel variables at the same time. This is done via GET and POST like the single get and set variable equivalents. Leading and trailing whitespace will be stripped from the variable names for both paths. When setting variables, the values will be read as-is, whitespace included. GET takes in a single string with comma-separated values, while POST takes in a dictionary of key value pairs. The code follows the same paths as when setting multiple variables when originating a channel via ARI. UserNote: Added new ARI paths for getting and setting multiple channel variables at a time. For GET, this takes in a single string of comma-separated variable names, while POST takes in a dictionary of key value pairs. The behavior is the same as passing in variables when originating a channel.	4 months ago
Bernd Kuhls	ba98ceae97	res_stir_shaken: avoid direct ASN1_STRING accesses https://github.com/openssl/openssl/issues/29117 Signed-off-by: Bernd Kuhls <bernd@kuhls.net> Resolves: #1952	4 months ago
Bernd Kuhls	9d608b50c1	tcptls.c: fix build with OpenSSL 4 tcptls.c: In function '__ssl_setup': tcptls.c:417:52: error: implicit declaration of function 'SSLv3_client_method'; did you mean 'SSLv23_client_method'? [-Wimplicit-function-declaration] 417 \| cfg->ssl_ctx = SSL_CTX_new(SSLv3_client_method()); SSLv3_client_method was removed from OpenSSL 4.0.0: https://github.com/openssl/openssl/blob/openssl-4.0.0/doc/man7/ossl-removed-api.pod?plain=1#L440 Signed-off-by: Bernd Kuhls <bernd@kuhls.net> Resolves: #1952	4 months ago
mikhail_grishak	b94728ece7	res_calendar: Fix build with libical 4.X libical 4.0 removed the icaltime_add() function in favor of icaltime_adjust(). Additionally, the callback signature for icalcomponent_foreach_recurrence() was updated to use a const pointer for the icaltime_span argument. This commit adds conditional compilation using ICAL_MAJOR_VERSION to support both libical 3.X and the new 4.X API, ensuring backward compatibility. Fixes: #1957	4 months ago
UpBeta	82ca1727c7	app_record: Fix hangup handling during beep playback When a hangup occurs while app_record is playing the initial beep, the application does not detect the hangup and continues running until the maxduration timeout expires. Replace the manual ast_streamfile() + ast_waitstream() sequence with ast_stream_and_wait(), which properly detects hangup and returns non-zero, allowing the application to exit immediately with RECORD_STATUS set to HANGUP. Resolves: #1950	4 months ago

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