asterisk

Commit Graph

Author	SHA1	Message	Date
Milan Kyselica	8cb6dfd41d	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	4 months ago
Milan Kyselica	e6998409f7	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	4 months ago
Milan Kyselica	9b085d48fb	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	4 months ago
Milan Kyselica	0e6d761ed8	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	4 months ago
Asterisk Development Team	a488f7dbd2	Update for 20.20.0	5 months ago
Asterisk Development Team	8110702bd1	Update for 20.20.0-rc1	5 months ago
George Joseph	8b12c95721	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	5 months ago
Mike Bradeen	b356d182d4	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]}`	5 months ago
Ben Ford	9b8143c296	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.	5 months ago
Bernd Kuhls	1f433ffec3	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	5 months ago
Bernd Kuhls	beb2623a0a	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	5 months ago
mikhail_grishak	a8b7a80a90	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	5 months ago
UpBeta	427dbc13e2	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	5 months ago
smtcbn	adfe662835	odbc: Don't use prepared statements for distinct SQL statements Avoids unnecessary prepare for simple INSERT statements that cause issues with ProxySQL (prepared statement counter overflow). Resolves: #1217	5 months ago
Alexander Bakker	34153dc07c	abstract_jb.c: Remove timerfd from channel when disabling jitter buffer Previously, the lingering timerfd would cause a tight loop if the channel enters a BridgeWait after the jitter buffer was disabled. Fixes: #1762	5 months ago
Sean Bright	5cd88cb64d	res_pjsip: Don't allow a leading period when wildcard matching The reference identifier (what the client provides - in this case a hostname) must start with a domain label, not a `.`. The current implementation will match `.seanbright.com` against `*.seanbright.com` which is incorrect.	5 months ago
George Joseph	1982afca0d	Ensure channel locks aren't held while calling ast_set_variables. If the channel is locked when calling ast_set_variables and any of the variables contained dialplan functions, there's a possiblilty of a deadlock. To prevent this, either the explicit locks were removed or the call to ast_set_variables moved out of the lock scope. A warning to not hold channel locks is also added to the documentation for ast_set_variables. Resolves: #1936	5 months ago
smtcbn	2a64ca82e6	app_queue: fix double increment of member->calls with shared_lastcall Under high concurrency, update_queue() may be invoked multiple times for the same call, causing member->calls and queue-level counters to be incremented more than once. The existing starttime check is not atomic and allows concurrent execution paths to pass. Treat member->starttime as a single-use token and consume it via CAS to ensure the call is counted exactly once. This also prevents incorrect call distribution when using strategies such as fewestcalls. Observed as a regression after upgrading to 20.17. Resolves: #1736	5 months ago
George Joseph	d380df0231	chan_dahdi: Fix set but not used in mfcr2_show_links_of(). When openr2 is installed mfcr2_show_links_of() is no longer ifdeffed out which makes gcc-16 complain with 'variable ‘x’ set but not used'. Resolves: #1947	5 months ago
Sebastian Jennen	fdaf266cd3	tests: add tests/test_codec_translations.c This tests checks [slin -> codec -> slin] and then compares slin in vs out regarding signal noise ratio and delay. Near-lossless codecs (ulaw, alaw) are checked with a maximum per-sample error bound. Lossy codecs are checked with a per-codec SNR threshold. Cross-correlation alignment compensates for algorithmic delay in codecs like speex and opus. Covered codecs: ulaw, alaw, adpcm, g726, g726aal2, gsm, speex, speex16, speex32, ilbc, codec2, lpc10, g722, opus. Resolves: #1812	5 months ago
Sean Bright	70e8a02f69	install_prereq: Add a 'minimal' mode for basic build dependencies	5 months ago
George Joseph	0a0542533d	chan_websocket: Handle incoming CONTINUATION frames. chan_websocket now tells res_http_websocket to accumulate incoming CONTINUATION frames into 1024 byte TEXT or BINARY frames. Resolves: #1941	5 months ago
George Joseph	a4b0ee20db	res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). ``` AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ rtp->rtcp->stdev_rxjitter); ``` Should have been ``` AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, \ AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, \ rtp->rtcp->stdev_rxmes); ``` Note the last macro parameter name. Resolves: #1938	5 months ago
Stanislav Abramenkov	6ac2a4824e	jansson: Upgrade version to jansson 2.15.0 UpgradeNote: jansson has been upgraded to 2.15.0. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0 Resolves: #1931	5 months ago
George Joseph	ac1cee069f	channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. The original trigger for setting the RTP stats in ast_softhangup() came from an ARI issue where stats weren't being set in time to be reported on STASIS_END events. The thought was that setting them in a common place like ast_softhangup() would ensure the stats were set in possibly other scenarios. Unfortunately, setting the RTP stats variables in ast_softhangup() broke ABI as it required that no channel locks be held which was not the case earlier. Given that the original issue was ARI, we can move setting the stats to ast_ari_channels_hangup() in resource_channels just before it calls ast_softhangup(). This might not catch all cases of the stats not being set, but it won't break ABI or deadlock either. Resolves: #1928	5 months ago
George Joseph	20de9cbbe5	res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution is enabled based on the TTL returned in the DNS results. If the TTL returned is 0, it means that the next time the IP address is needed, it must be looked up again. I.E. Don't cache. Historically (and incorrectly) however, res_rtp_asterisk stopped the periodic resolution and never re-resolved the hostname again. Besides what's mentioned in the user notes... * Additional debugging was added in various STUN/DNS functions. * The `rtp show settings` CLI command shows more detailed STUN info. * Some debugging was added to dns_core.c and dns_recurring.c. UserNote: A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf that allows control over what happens when a STUN server hostname lookup returns a TTL of 0. The values can be set as follows: - 'no': This is the historical (and current default) behavior of not doing any further lookups and continuing to use the last successful result until Asterisk is restarted or rtp.conf is reloaded. - 'yes': Use the last cached result for the current call but trigger re-resolution in the background for the benefit of future calls. If the result of the background lookup is a ttl > 0, periodic resolution will be restarted otherwise the next call will use the new cached value and will trigger a background lookup again. UserNote: A new CLI command `rtp resolve stun hostname` has been added that will force a resolution of the STUN hostname and (re)start periodic resolution if the result has a TTL > 0. Resolves: #1858	5 months ago
Jaco Kroon	a77095817a	pjsip_configuration: Show actual dtls_verify config. Rather than merely showing dtls_verify : Yes/No in pjsip show endpoint xxx it will now be shown what exactly is being checked, ie, one of: dtls_verify : No dtls_verify : Fingerprint dtls_verify : Certificate dtls_verify : Yes Where Yes implies both Fingerprint and Certificate. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 months ago
Naveen Albert	2a95cd54f4	app_dial: Properly handle callee hangup while sending digits. If we are sending digits (either DTMF, MF, or SF) to the called channel after receiving progress or a wink, and the callee hangs up before we have finished sending it digits, there are several problems that can ensue: * If the callee hung up without answering, the calling channel would hang up and not continue in the dialplan. * If the callee did answer before hanging up, the answer was never passed through to the caller, since this gets "eaten" by the various digit streaming functions and is never processed by app_dial. This is generally an edge case that occurs due to some kind of signaling failure, but to better handle this: * Set to_answer to 0 to prevent hangup on the exit path, just like other parts of wait_for_answer. * Better document this usage of to_answer. * If the channel did answer while it was receiving digits, manually answer the calling channel before we abort. The call would not continue in the dialplan anyways (either before or after this fix), but technically the call was answered, so the CDRs should probably reflect that, and this mirrors the behavior of calls which normally do not continue. Resolves: #1915 UserNote: If a called channel sends progress or wink and the caller begins sending digits but the callee answers and then hangs up before digit sending can finish, the call is now answered before being disconnected. If the callee hangs up without answering, the call now continues in the dialplan.	5 months ago
Maximilian Fridrich	ea8ecd6dea	res_pjsip_messaging: Update To URI only if it is a SIP(S) URI When a message is sent via ARI, the ARI endpoint only provides a To field which is also used as destination field. This means that the To field might not necessarily contain a SIP URI but might instead specify an Asterisk endpoint (in MessageDestinationInfo format). This led to many warnings even though the message was sent correctly. The fix is to only call `ast_sip_update_to_uri` if the To field starts with the sip: or sips: scheme. Resolves: #1357	5 months ago
Stanislav Abramenkov	a49b33e74a	Upgrade bundled pjproject to 2.17. Resolves: #1888 UserNote: Bundled pjproject has been upgraded to 2.17. For more information about what is included in this release, see the pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17	5 months ago
Mike Bradeen	e90e3ea403	res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug crypto_utils uses ast_asprintf to allocate the search string when checking the certificate subject, but was not using ast_free to free it. This caused a crash when Asterisk was built with malloc_debug Resolves: #1921	5 months ago
Joshua C. Colp	40cc2748d0	manager: Eliminate unnecessary code, simplify sessions in stasis callbacks Due to stasis filtering the stasis callback for AMI type messages is guaranteed to only receive messages that can be turned into AMI events, so remove the check done in the callback. The sessions container usage for the stasis callbacks has also been simplified by having a reference on the message router subscription instead of having to acquire the sessions from the global object each time.	5 months ago
Peter Krall	f11c71ca36	res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup Modified the bridge playback teardown so the worker thread removes only the playback control, while the after-bridge callback removes the playback wrapper once the announcer has actually left the bridge. This avoids a stale window where a new playback request could create a replacement announcer before the old announcer had fully exited the holding bridge. Also replaced the flexible trailing bridge_id storage in the shared worker thread data with an optional bridge_id pointer, since recording paths use the same structure without a bridge id. Fixes: #1861	5 months ago
Sebastian Denz	c6421d8143	res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage	5 months ago
George Joseph	9ed0a3c38a	channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars ast_softhangup() was locking the channel before calling ast_rtp_instance_set_stats_vars() which, if the channel was in a bridge, then locked the bridge peer channel. If another thread attempted to set bridge variables on the peer, it would lock that channel first, then this channel causing a lock inversion. ast_softhangup() now holds the channel lock while retrieving the rtp instance, then unlocks it before calling ast_rtp_instance_set_stats_vars(), then locks it again after it returns. Resolves: #1907	5 months ago
Charles Langlois	cd5aa39670	chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER When a PJSIP endpoint is configured with set_var invoking a dialplan function (e.g. PJSIP_HEADER(add,...)), chan_pjsip_new() calls pbx_builtin_setvar_helper() while holding the channel lock. For function-style variables, this dispatches to ast_func_write() which, in the case of PJSIP_HEADER, calls ast_sip_push_task_wait_serializer() -- blocking synchronously while the channel lock is held. If a concurrent operation (ARI, AMI, rtp_check_timeout) traverses the channels container via ast_channel_get_by_name(), it acquires the container lock then tries to lock individual channels in the iteration callback (by_uniqueid_cb/by_name_cb). When the serializer thread also needs the container lock, a circular dependency forms: channel_lock -> serializer_wait -> container_lock -> channel_lock This causes a complete Asterisk freeze. In the observed case, 36 threads were blocked on the container lock until res_freeze_check triggered SIGABRT after its 30-second timeout. Unlock the channel before iterating endpoint channel_vars so that dialplan functions can block without holding the channel lock. Re-lock the channel for ast_channel_stage_snapshot_done() so the batched snapshot is published under lock and captures the full channel state including the variables set during the loop. Fixes: #1872	5 months ago
mattia	9da9d57d5e	res_pjsip: Add per-endpoint RTP port range configuration Add rtp_port_start and rtp_port_end options to PJSIP endpoint configuration, allowing each endpoint to use a dedicated RTP port range instead of the global rtp.conf setting. This is useful for scenarios where different endpoints need isolated port ranges, such as firewall rules per trunk, multi-tenant systems, or network QoS policies tied to port ranges. The implementation adds ast_rtp_instance_new_with_port_range() to the RTP engine API, which sets the port range on the instance before the engine allocates the transport. The default RTP engine (res_rtp_asterisk) checks for per-instance overrides in rtp_allocate_transport() and falls back to the global range when none is set. Both options must be set together, with values >= 1024 and rtp_port_end > rtp_port_start. Setting both to 0 (the default) preserves existing behavior. Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71 UserNote: PJSIP endpoints now support rtp_port_start and rtp_port_end options to configure a dedicated RTP port range per endpoint, overriding the global rtp.conf setting. UpgradeNote: An alembic database migration has been added to add the rtp_port_start and rtp_port_end columns to the ps_endpoints table. Run "alembic upgrade head" to apply the schema change. DeveloperNote: New public API: ast_rtp_instance_new_with_port_range() creates an RTP instance with a per-instance port range. ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() allow RTP engines to query the override. Third-party RTP engines can use these getters to support per-instance port ranges.	5 months ago
phoneben	7655a02349	app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules app_queue: Fix raise_respect_min not copied in copy_rules() causing rN rules to be ignored. `copy_rules()` never copied `raise_respect_min` into the per-call rule list, so the flag was always 0 when a timed penaltychange rule fired, making `rN` behave like plain `N` and raising members below `min_penalty` that should have been excluded. Also fixes `update_qe_rule()` not propagating the flag from `qe->pr` to `qe`, and dropping the `r` prefix when saving back to `QUEUE_RAISE_PENALTY`. Resolves: #1901	5 months ago
phoneben	f9cb6f8bae	app_voicemail_odbc: fix msgnum race and crash on failed STORE app_voicemail_odbc: fix msgnum race and crash on failed STORE Two concurrent callers leaving voicemail to the same mailbox could be assigned the same msgnum because ast_unlock_path() was called before STORE(), allowing a second thread to read the same LAST_MSG_INDEX() before the first INSERT committed. The losing thread got a duplicate key error, but execution continued into notify_new_message() -> RETRIEVE() because the STORE() return value was not checked. RETRIEVE() then fetched the winning thread's DB row, mmap'd its blob size against the locally truncated file, and crashed with SIGBUS. Hold the path lock through STORE() and bail out on failure. Fixes: #1653	5 months ago
George Joseph	e247f1f24f	ari_websockets: Fix two issues in the cleanup of outbound websockets. 1. session_cleanup() now saves the websocket type before unlinking the session from the session registry. This prevents a FRACK when cleaning up per-call websockets when MALLOC_DEBUG is used. 2. session_shutdown_cb() and outbound_sessions_load() now call pthread_cancel() to cancel the session handler thread to prevent the thread from continually trying to connect to a server after the connection config has been removed by a reload. This required the thread to use pthread_cleanup_push() to clean up its reference to the session instead of RAII because RAII destructors don't get run when pthread_cancel() is used. Resolves: #1894	5 months ago
George Joseph	93933db9cc	compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++. `__STDC_VERSION__` is specific to C but up until gcc 16, the g++ compiler also defined it. With g++ 16.0 it's no longer defined (which is the correct behavior) so compiling channelstorage_cpp_map_name_id.cc fails. The check for `__STDC_VERSION__` in compat.h is now skipped if we're compiling a C++ source file. Resolves: #1903	5 months ago
phoneben	4346b430ca	pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c Backport pjsip/pjproject#4941 which fixes a build/link failure when compiling against OpenSSL < 1.1.0 (e.g. OpenSSL 1.0.2k on CentOS 7). Two symbols introduced in OpenSSL 1.1.x were called unconditionally in ssl_sock_ossl.c without version guards: - `TLS_method()` in `init_ossl_ctx()` is now guarded with `OPENSSL_VERSION_NUMBER < 0x10100000L`, falling back to `SSLv23_method()` on older OpenSSL. - `SSL_CTX_set_ciphersuites()` is now guarded with `OPENSSL_VERSION_NUMBER >= 0x1010100fL` since this function was introduced in OpenSSL 1.1.1 and is absent in 1.0.x. Without this fix, linking fails with: undefined reference to `TLS_method' undefined reference to `SSL_CTX_set_ciphersuites' when building Asterisk with bundled pjproject on systems such as CentOS 7 with OpenSSL 1.0.2k. Resolves: #1892	5 months ago
George Joseph	e9cca45141	asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. Line 2729 has `#if HAVE_LIBEDIT_IS_UNICODE` instead if `#ifdef`. Since macros defined by autoconf are either set to `1` or not set at all, older distros where libedit isn't unicode won't have that macro defined and will fail to compile. Resolves: #1896	5 months ago
phoneben	7116b76c06	cdrel_custom: fix SQLite compatibility for versions < 3.20.0 cdrel_custom: fix SQLite compatibility for versions < 3.20.0 Replace sqlite3_prepare_v3 + SQLITE_PREPARE_PERSISTENT with a version-guarded fallback to sqlite3_prepare_v2 for older SQLite builds. Resolves: #1885	5 months ago
Sebastian Jennen	719170e7f3	translate.c: implement different sample_types for translation computation. The default (codec) still uses the codec provided samples. Additionally different sample_types can be used with eg: `translate sampletype speech` and then running `core show translation comp 10` to measure performance of different audio scenarios. Resolves: #1807	5 months ago
Daniel Donoghue	63cf60c43c	stasis_broadcast: Add optional ARI broadcast with first-claim-wins Adds two optional modules: res_stasis_broadcast.so: Infrastructure for broadcasting a single incoming channel to multiple ARI applications with atomic first-claim-wins semantics. app_stasis_broadcast.so: Provides the StasisBroadcast() dialplan application which invokes the broadcast infrastructure. Both modules are self-contained; if neither is loaded there is zero runtime impact. Loading them does not alter existing Stasis or ARI behavior unless explicitly used. Key Features (only active when modules are loaded): Fisher-Yates shuffled broadcast dispatch for fair claim races Atomic claim operations using mutex + condition variable signaling Configurable broadcast timeouts Safe regex application filtering with validation to mitigate ReDoS risk Thread-safe channel variable snapshotting (channel locked during reads) Late-claim safety: broadcast context kept alive until after the Stasis session ends so concurrent claimants always receive 409 Conflict rather than 404 Not Found Memory safety via RAII_VAR, ast_json_ref/unref, and ao2 reference counting Components Added: res/res_stasis_broadcast.c: Core broadcast + claim logic apps/app_stasis_broadcast.c: StasisBroadcast() dialplan application include/asterisk/stasis_app_broadcast.h: Public API header res/ari/resource_events.c: Integrates POST /ari/events/claim endpoint rest-api/api-docs/events.json: New CallBroadcast and CallClaimed events Implementation Notes: Broadcast contexts reside in an ao2 hash container keyed by channel id. Each context holds atomic claim state, winner application name, timeout metadata, and a condition variable for waiters. Broadcast contexts are kept alive until after stasis_app_exec() returns so that concurrent claimants racing against the timeout always receive 409 Conflict. Broadcast dispatch calls stasis_app_send() directly for each matching application in shuffled order. Regex filters are validated with bounded length, group depth, quantified group count, and alternation limits to reduce pathological backtracking. Timeout calculation uses timespec arithmetic with overflow-safe millisecond remainder handling. Event JSON follows existing Stasis/ARI conventions; references are managed correctly to avoid leaks or double frees. Optional Nature / Impact: No changes to existing APIs, events, or applications when absent. Clean fallback: systems ignoring the modules behave identically to prior versions. Development was assisted by Claude (Anthropic). All generated code has been reviewed, tested, and is understood by the author. UserNote: New optional modules res_stasis_broadcast.so and app_stasis_broadcast.so enable broadcasting an incoming channel to multiple ARI applications. The first application to successfully claim (via POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan application initiates broadcasts. CallBroadcast and CallClaimed events notify applications. When modules are not loaded, behavior is unchanged. DeveloperNote: New public APIs in stasis_app_broadcast.h: stasis_app_broadcast_channel(), stasis_app_claim_channel(), stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; no existing ABI modified.	5 months ago
Sven Kube	776a71e6d4	res_audiosocket: Tolerate non-audio frame types This commit implements the handling of non-voice or DTMF frames like the chan_websocket handling added in #1588. Rather than treating unsupported frames as fatal errors, silently ignore CNG frames and log a warning for other unsupported types.	5 months ago
George Joseph	37118bb860	pbx_functions: Save module pointer before calling read and write callbacks. Before ast_func_read and ast_func_write call their respective read and write callbacks for registered dialplan functions, they use the module pointer in the registered ast_custom_function structure to increment the module use count. They then decrement the usecount when the callback returns. This prevents the providing module from being unloaded while there's a call using the function. Some modules, notably func_odbc, create and destroy dialplan functions based on the contents of a config file. Since the ast_custom_function structure is dynamically allocated, it could be destroyed on reload which means when the module's read or write callback returns to the ast_func calls it would try to decrement the usecount using the module pointer from an ast_custom_function structure that has already been freed. Proper locking or reference counting by the module can reduce the possibility of this happening but it can't prevent it because it doesn't have control after its read or write callback has returned to ast_func_read or ast_func_write. To address this, ast_func_read, ast_func_read2 and ast_func_write save the module pointer to a local variable before calling the module's callback, then use the saved pointer to decrement the use count. The module pointer will always be valid if the module is loaded regardless of the state of the ast_custom_function structure. Resolves: #1818	5 months ago
Naveen Albert	6d322b5658	chan_iax2: Add CHANNEL getter to retrieve auth method. Add a property to the CHANNEL method to retrieve the auth method, which can be used to retrieve the specific auth method actually negotiated for a call (e.g. RSA, MD5, etc.). Also clean up some of the documentation for the secure properties to clarify how these relate to call encryption. Resolves: #1878 UserNote: CHANNEL(auth_method) can now be used to retrieve the auth method negotiated for a call on IAX2 channels.	5 months ago
phoneben	75c1f04be0	fix: backport pjproject stdatomic.h GCC 4.8 build failure patch pjproject 2.16 (bundled) fails to build on GCC 4.8 (CentOS/RHEL 7) due to a false positive C11 atomics detection introduced in pjproject commit #4570. A fix has been submitted upstream to pjproject (#4933). Adding a local patch to third-party/pjproject/patches/ until a fixed version of pjproject is bundled in Asterisk. Fixes build error: ../src/pj/os_core_unix.c:52:27: fatal error: stdatomic.h: No such file or directory Resolves: #1883	5 months ago

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