34405 Commits (7988d1138cb2dfd27e3e18f4af0672618f67eb6d)

 

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Author	SHA1	Message	Date
Naveen Albert	7988d1138c	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.	4 months ago
Maximilian Fridrich	f4ed6a13fc	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	4 months ago
Stanislav Abramenkov	4ec7509d93	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	4 months ago
Mike Bradeen	a95b21c556	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	4 months ago
Joshua C. Colp	f3a64f705e	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.	4 months ago
Peter Krall	176ecdad3c	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	4 months ago
Sebastian Denz	d94025d5df	res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage	4 months ago
George Joseph	3e52d75281	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	4 months ago
Charles Langlois	f616affc2a	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	fcb98380e4	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	cef63b761f	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	9e5f769e80	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	f7b60322b1	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	7a57d9f2ed	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	d9b8066ec0	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	590b490ba2	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	48e82dec45	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	4b9ef3f5e2	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	4413a02d0c	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	97798a0554	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	a899013d56	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	6af0e04001	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	1b932f188a	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
George Joseph	a5b77874a3	res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy. ... The rtp_ioqueue_thread_destroy() function was destroying the the ioqueue thread and releasing its pool but not destroying the ioqueue itself. This was causing the ioqueue's epoll file descriptor to leak. Resolves: #1867	5 months ago
Daniel Donoghue	b3728145b2	res_pjsip_maintenance: Add PJSIP endpoint maintenance mode ... Introduces res_pjsip_maintenance, a loadable module that allows operators to place individual PJSIP endpoints into maintenance mode at runtime without unregistering or disabling them. While an endpoint is in maintenance mode: * New inbound INVITE and SUBSCRIBE dialogs are rejected with 503 Service Unavailable and a Retry-After: 300 header. * In-progress dialogs (re-INVITE, UPDATE, BYE, etc.) are unaffected and complete normally. * Outbound originations via Dial() or ARI originate are refused before any SIP session is created. State is held in-memory only and is cleared on module unload or Asterisk restart. This module was developed with AI assistance (Claude). All code has been reviewed and tested by the author, who takes full responsibility for the submission. CLI interface: pjsip set maintenance <on|off> <endpoint|all> pjsip show maintenance [endpoint] AMI interface: Action: PJSIPSetMaintenance Endpoint: <name>|all State: on|off Action: PJSIPShowMaintenance Endpoint: <name> (optional; omit to list all) Emits PJSIPMaintenanceStatus events per result, followed by PJSIPMaintenanceStatusComplete. State changes also emit an unsolicited PJSIPMaintenanceStatus event. To support outbound blocking, a new session_create callback is added to ast_sip_session_supplement. Supplements that set this callback are invoked at the start of ast_sip_session_create_outgoing() in res_pjsip_session, before any dialog or invite session resources are allocated. res_pjsip_maintenance registers itself as a session supplement and uses this callback to gate outbound session creation on a per-endpoint basis. MODULEINFO: <depend>pjproject</depend> <depend>res_pjsip</depend> <depend>res_pjsip_session</depend> UserNote: New module res_pjsip_maintenance adds runtime maintenance mode for PJSIP endpoints. Use "pjsip set maintenance <on|off> <endpoint|all>" to enable or disable, and "pjsip show maintenance" to list affected endpoints. AMI actions PJSIPSetMaintenance and PJSIPShowMaintenance provide programmatic access. No configuration file changes required. DeveloperNote: ast_sip_session_supplement gains a new optional callback - int (*session_create)(struct ast_sip_endpoint *endpoint, const char *destination). It is called from the global supplement list (not per-session) at the start of ast_sip_session_create_outgoing() via ast_sip_session_check_supplement_create(). Returning non-zero blocks the outgoing session. Modules that need to gate outbound SIP session creation should register a supplement with this callback set rather than hooking into chan_pjsip directly.	5 months ago
Naveen Albert	ce9765032f	chan_iax2: Add another check to abort frame handling if datalen < 0. ... Commit 2da221e217 added a missing abort if datalen < 0 check on a code path and an assertion inside iax_frame_wrap if we ever encountered a frame with a negative frame length (which will eventually cause a crash). Add another missing abort check for negative datalen, exposed by this assertion. (Similar to the previous commit, this is a video frame with a datalen of -1). Resolves: #1865	5 months ago
Mike Bradeen	8afbd78139	res_pjsip_outbound_registration: only update the Expires header if the value has changed ... The PJSIP outbound registration API has undocumented behavior when reconfiguring the outbound registration if the expires value being set is the same as what was previously set. In this case PJSIP will remove the Expires header entirely from subsequent outbound REGISTER requests. To eliminate this as an issue we now check the current expires value against the configured expires value and only apply it if it differs. This ensures that outbound REGISTER requests always contain an Expires header. Resolves: #1859	5 months ago
Sean Bright	5e837f1039	func_talkdetect.c: Clarify dsp_talking_threshold documentation. ... Fixes: #1761	5 months ago
Sean Bright	c3fef35ad2	make_xml_documentation: Remove temporary file on script exit. ... Fixes: #1862	5 months ago
George Joseph	006e25471d	res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread ... When res_pjsip is reloaded directly, it does the sorcery reload in a pjsip servant thread as it's supposed to. res_pjsip_config_wizard however was not which was leading to occasional deadlocks. It now does the reload in a servant thread just like res_pjsip. Resolves: #1855	5 months ago
Alexei Gradinari	47011850d4	build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build ... The pjsua Python module and the pjsua/pjsystest apps were used by the Asterisk Test Suite for SIP simulation in dev mode builds. They are now fully obsolete for three independent reasons: 1. **pjsua Python bindings officially deprecated upstream.** The pjproject maintainers added `pjsip-apps/src/python/DEPRECATED.txt` directing users to the PJSUA2 SWIG binding instead. A build-fix PR (https://github.com/pjsip/pjproject/pull/4892) was closed by the maintainer explicitly citing this deprecation. 2. **Removed from the Asterisk Test Suite.** As confirmed by @mbradeen: > *"We had to get rid of pjsua when we went to Python3 because it would > hang due to a conflict between async calls within pjsua and twisted. > There are still some old references to tests we couldn't fully convert > to sipp, but those are skipped."* 3. **Broken and unmaintained.** Building with Python 2.7 (the only version `configure.ac` searched for) fails with: ``` _pjsua.c: error: 'INIT_RETURN' undeclared (first use in this function) ``` due to a bug in pjproject 2.16's `python3_compat.h` that upstream declined to fix. This PR removes all pjsua-related build artifacts from Asterisk's bundled pjproject build: the pjsua and pjsystest application binaries, the deprecated Python (`_pjsua.so`) bindings, the `asterisk_malloc_debug.c` stubs, and the `PYTHONDEV` detection from `configure.ac`. Also removes `libpjsua` from Asterisk's main linker flags. DeveloperNote: The pjsua and pjsystest application binaries, the deprecated Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub implementations are no longer built or installed as part of the bundled pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency is also removed. Developers who relied on the pjsua binary for Test Suite SIP simulation should use SIPp instead, which is the current Asterisk Test Suite standard. Fixes: #1840	5 months ago
nappsoft	3dd86beeac	res_cdrel_custom: do not free config when no new config was loaded ... When the res_cdrel_custom modules is reloaded and the config has not been changed asterisk should not free the old config. Otherwise the connection to the database will be closed and no new connection will be opened. Resolves: #1852	5 months ago
George Joseph	80b953f76f	res_cdrel_custom: Resolve several formatting issues. ... Several issues are resolved: * Internally, floats were used for timestamp values but this could result in wrapping so they've been changed to doubles. * Historically, the default CEL eventtime format is `<seconds>.<microseconds>` with `<microseconds>` always being 6 digits. This should have continued to be the case but res_cdrel_custom wasn't checking the `dateformat` setting in cel.conf and was defaulting to `%F %T`. res_cdrel_custom now gets the default date format from cel.conf, which will be whatever the `dateformat` parameter is set to or `<seconds>.<microseconds>` if not set. * The timeval field formatter for both CDR and CEL wasn't handling custom strftime format strings correctly. This is now fixed so you should be able to specifiy custom strftime format strings for the CEL `eventtime` and CDR `start`, `answer` and `end` fields. For example: `eventtime(%FT%T%z)`. Resolves: #1844 Resolves: #1845	5 months ago
Milan Kyselica	3e0817d8e6	callerid: fix signed char causing crash in MDMF parser ... Change rawdata buffer from char to unsigned char to prevent sign-extension of TLV length bytes >= 0x80. On signed-char platforms (all Asterisk builds due to -fsigned-char in configure.ac), these values become negative when assigned to int, bypass the `if (res > 32)` bounds check, and reach memcpy as size_t producing a ~18 EB read that immediately crashes with SIGSEGV. Affects DAHDI analog (FXO) channels only. Not reachable via SIP, PRI/BRI, or DTMF-based Caller ID. Fixes: #1839	5 months ago
Mike Bradeen	500525c289	res_pjsip: Address pjproject security vulnerabilities ... Address the following pjproject security vulnerabilities [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp) [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc) [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7) [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj) Resolves: #1833	6 months ago
Tinet-mucw	2f85c76fb9	pbx: Hold channel lock for exception datastore access ... ast_channel_datastore_find() and ast_channel_datastore_add() must only be called while the channel is locked (see channel.h). raise_exception() and the EXCEPTION dialplan function read path accessed the exception datastore without holding ast_channel_lock, which could corrupt the per-channel datastore list under concurrency and lead to crashes during teardown (e.g. double free in ast_datastore_free). Resolves: #1831	6 months ago
George Joseph	fc931fcba8	xmldoc.c: Fix memory leaks in handling of provided_by. ... Added a few calls to ast_xml_free_attr() to squash memory leaks when handling "provided_by".	6 months ago
George Joseph	5a7b6cf1c3	SECURITY.md: Update with additional instructions. ... Also added line breaks for people reading this file directly from the code base.	6 months ago
Sven Kube	4ba7014f89	res_audiosocket: Fix header read loop to use correct buffer offset ... The PR #1522 introduced the header read loop for audiosocket packets which does not handle partial header reads correctly. This commit adds the missing buffer offsets.	6 months ago
phoneben	ef9893d520	manager.c : Fix CLI event display ... manager.c: Fix CLI event display - `manager show events`: fix event names being truncated at 20 characters, widen column to 28 to accommodate the longest registered event name - `manager show events`: skip duplicate entries caused by multiple modules registering the same event name, list is already sorted so adjacent name comparison is sufficient	6 months ago
George Joseph	5084177cb2	chan_pjsip: Set correct cause codes for non-2XX responses. ... Redirects initiated by 302 response codes weren't handled correctly when setting the hangup cause code and tech cause code on the responding channel. They're now set to 23 (REDIRECTED_TO_NEW_DESTINATION) and 302 (Moved permanently). Other non-2XX response codes also had issues. A new API ast_channel_dialed_causes_iterator() was added to retrieve the hangup cause codes for a channel. chan_pjsip_session_end() in chan_pjsip has been refactored to set the correct cause codes on a channel based on the cause codes added by chan_pjsip_incoming_response_update_cause(). Copious amounts of debugging and comments were also added. Resolves: #1819	6 months ago
Michal Hajek	7210a1b4c2	res_pjsip_config_wizard: Force reload on Named ACL change events ... Currently, endpoints created via the PJSIP Config Wizard do not update their ACL rules if the underlying Named ACL (in acl.conf) changes. This occurs because the wizard relies on file timestamp and content caching of pjsip_wizard.conf, which remains unchanged during an external ACL update. As a result, endpoints retain stale ACL rules even after a reload. This patch updates res_pjsip_config_wizard to subscribe to the ast_named_acl_change_type Stasis event. A local generation counter is incremented whenever an ACL change event is received. During a reload, the wizard compares the current local generation against the generation stored in the wizard object. If a change is detected: 1. The file cache optimization (CONFIG_FLAG_FILEUNCHANGED) is bypassed. 2. Wizard objects utilizing 'acl' or 'contact_acl' are forced to update, ensuring they pick up the new IP rules. Signed-off-by: Michal Hajek michal.hajek@daktela.com Fixes: #1641	6 months ago
George Joseph	c2ee154ed2	rtp: Set RTPAUDIOQOS variables when ast_softhangup is called. ... If a channel in Stasis/ARI is hung up by the channel driver, the RTPAUDIOQOS variables are set before the channel leaves Stasis and are therefore available to the ARI app via ChannelVarset events. If the channel is hung up by ARI however, the channel leaves Stasis before the RTPAUDIOQOS variables are set so the app may not get the ChannelVarset events. We now set the RTPAUDIOQOS variables when ast_softhangup() is called as well as when the channel driver hangs up a channel. Since ARI hangups call ast_softhangup(), the variables will be set before the channel leaves Stasis and the app should get the ChannelVarset events. ast_rtp_instance_set_stats_vars(), which actually sets the variables, now checks to see if the variables are already set before attempting to set them. This prevents double messages from being generated. Resolves: #1816	6 months ago
Alexis Hadjisotiriou	7b8339fb52	channel: Prevent crash during DTMF emulation when no timing module is loaded ... Description: When Asterisk is running without a timing module, attempting to process DTMF triggers a segmentation fault. This occurs because the system attempts to access a null timing file descriptor when setting up the DTMF emulation timer. This fix ensures that the system checks for a valid timing source before attempting to start the DTMF emulation timer. If no timing module is present, it logs a warning and skips the emulation instead of crashing the process. Changes: - Modified main/channel.c to add a safety check within the __ast_read function. - Implemented a graceful return path when no timing source is available - Added a LOG_WARNING to inform the administrator that DTMF emulation was skipped due to missing timing modules. Testing: - Disabled all timing_ modules in modules.conf and confirmed with 'timing test'. - Reproduced the crash by modifying the dialplan with: exten => 707,1,NoOp(Starting DTMF - No Timing Mode) same => n,Answer() same => n,Background(demo-congrats) same => n,WaitExten(10) same => n,Hangup() And calling 707 followed by 1 - Verified that with the fix applied, the system logs "No timing module loaded; skipping DTMF timer" and continues dialplan execution without crashing. - Confirmed stability during concurrent media sessions and DTMF input. Fixes: #566	6 months ago
George Joseph	5656a5851b	res_pjsip: Remove temp transport state when a transport fails to load. ... If a pjsip transport (A) fails to load, its temporary state gets left behind causing the next transport to load (B) to pick up some of its parameters, including its name. This can cause B to have the correct name (B) in its transport object but the wrong name (A) in its internal state object. When a transport state is searched for later on, transport state B is returned but a retrieval of the actual transport object will fail because B's transport state id is actually "A" and transport "A" doesn't exist because it failed to load. remove_temporary_state() is now being called in all error paths in config_transport.c functions that call find_or_create_temporary_state(). A bit of extra debugging was also added to res_pjsip_nat.c. Resolves: #1814	6 months ago
Alexis Hadjisotiriou	24f3941d5e	res_pjsip_messaging: Remove Contact header from out-of-dialog MESSAGE as per RFC3428 ... According to RFC 3428 (Section 5), a Contact header is not required in a MESSAGE request unless the sender wants to establish a session. This patch ensures that the Contact header is removed from out-of-dialog MESSAGE requests within res_pjsip_messaging.c. Fixes: #1356	6 months ago
Mike Bradeen	46292c9a45	acl: Add ACL support to http and ari ... Add uri prefix based acl support to the built in http server. This allows an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict access. Add user based acl support for ARI. This adds new acl options to the user section of ari.conf to restrict access on a per user basis. resolves: #1799 UserNote: A new section, type=restriction has been added to http.conf to allow an uri prefix based acl to be configured. See http.conf.sample for examples and more information. The user section of ari.conf can now contain an acl configuration to restrict users access. See ari.conf.sample for examples and more information	6 months ago
Robert Wilson	31666e4039	res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection occurs before ICE candidate check ... When TURN is configured in rtp.conf, pjproject re-injects TURN packets via 127.0.0.1 (the loopback address). The DTLS packet handler checks the source address against the ICE active candidate list before the loopback address substitution runs, causing the packet to be silently dropped as the source 127.0.0.1 is not in the candidate list. Fix by performing the loopback address substitution before the ICE candidate source check in the DTLS path, mirroring the logic already present in the non-DTLS RTP path. Fixes: #1795 UserNote: WebRTC calls using TURN configured in rtp.conf (turnaddr, turnusername, turnpassword) will now correctly complete DTLS/SRTP negotiation. Previously all DTLS packets were silently dropped due to the loopback re-injection address not being in the ICE active candidate list.	6 months ago
Naveen Albert	7d3fb10c02	dsp.c: Add support for detecting R2 signaling tones. ... Extend the existing DTMF/MF tone detection support by adding support for R2 tones, another variant of MF (R1) signaling. Both forward and backward signaling are supported. Resolves: #1804	6 months ago
Naveen Albert	83a215ccee	app_dial: Include channel name in warnings during wait_for_answer. ... Include the channel name in warnings during wait_for_answer to make them more useful and allow problematic channels to be easily identified. Resolves: #1802	6 months ago