4487 Commits (22.12.0-rc2)

Author	SHA1	Message	Date
Mike Bradeen	c45d43ec7e	Audiohooks: Fix whisper framehook timer leak and masquerade handling ... Resolves several issues with the framehook added to allow whispering without an underlying stream. The timer is now owned by the audiohook list rather than the framehook. It is retired on both channels during a masquerade and recreated when the audiohook(s) are moved. If the timer cannot be started, whisper audio is still mixed into the outbound media as it was previously (requiring an underlying stream.) An audiohook that cannot be moved to the new channel is now left DONE so that its owner does not wait on it forever. Resolves: #2134 DeveloperNote: ast_channel_fd_add() now returns -1 if it fails to allocate space for the descriptor.	7 days ago
Mike Bradeen	0b4cba4346	res_pjsip: Qualify all resolved contact targets ... Previously, PJSIP endpoint qualification was offloaded to pjproject which used a single, internally selected contact based on DNS results. This could result in unused SRV, A, or AAAA targets, leaving a contact marked as unreachable without probing the remaining targets. Now, we resolve the complete set of records before sending and creating OPTIONS transactions for every resolved target. The contact will be marked as reachable as soon as any target succeeds and unreachable only after every target fails. Fixes: #1927 UserNote: DNS-backed PJSIP contacts are now considered reachable when any resolved SRV, A, or AAAA target responds to qualification.	4 weeks ago
George Joseph	7d20b69a33	ARI, chan_websocket, res_websocket_client: Handle non-blocking and timeout correctly. ... This change was prompted by an issue where if the websocket peer to which ARI or chan_websocket is connected stops consuming TCP packets, its TCP receive buffer will begin to fill and ultimately cause the client's TCP send buffer to start filling. When the send buffer is completely filled, further writes to the websocket will block. This can cause a cascading lock situation with chan_websocket because it sends large amounts of data very quickly but can also affect ARI. * Added "write_timeout" parameters to websocket_client.conf and chan_websocket.conf. ari.conf already had the "websocket_write_timeout" parameter. See notes below. * Updated chan_websocket to use the write_timeout set in chan_websocket.conf for incoming/server connections if set and to use AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT (100ms) as the default if not set. * Updated chan_websocket to use the write_timeout set in websocket_client.conf for outgoing/client connections if set and to use write_timeout from chan_websocket.conf if not set. If neither is set, AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT (100ms) is used as the default. * Set non-blocking mode on the websocket in chan_websocket for both client and server connections. ARI already sets it. This is required for the timeouts to operate correctly. * Updated sample config files and XML documentation for all 3 areas. Resolves: #2140 UpgradeNote: A new "write_timeout" parameter has been added to websocket_client.conf that allows setting the maximum amount of time a write operation can take before returning an error. UpgradeNote: A new "write_timeout" parameter has been added to chan_websocket.conf setting the maximum amount of time a write operation can take before returning an error. Outgoing/client connections can override this with the new "write_timeout" parameter that was also added to websocket_client.conf. DeveloperNote: A new "write_timeout" field has been added to the ast_websocket_client_options structure that allows setting the maximum amount of time a write operation can take before returning an error. The websocket session must be set to nonblocking for this to take effect.	4 weeks ago
whitetreebug	75f2f59f5f	res_prometheus: add toggle config for metrics ... In high‑volume call scenarios, Prometheus exports a large number of channels‑related and bridges-related metrics.Not all users require channels_state, channels_duration and bridges_channels.Add conf options to enable or disable channel_details(covering channels_state, channels_duration) and bridge_details(covering bridges_channels). UserNote: Add new config option 'channels_detail_metrics_enabled' to disable export of 'asterisk_channels_state' and 'asterisk_channels_duration_seconds' metrics,It defaults to 'yes' to preserve existing behavior Add new config option 'bridges_detail_metrics_enabled' to disable export of 'asterisk_bridges_channels_count' metric, It defaults to 'yes' to preserve existing behavior Explicitly enable bridges_detail_metrics_enabled for the bridge_to_string unit‑test.	4 weeks ago
Jacky W J Li	9ff74ffbfb	frame: Correct ast_frame_adjust_volume* documentation of 'adjustment' ... The documentation for ast_frame_adjust_volume() and ast_frame_adjust_volume_float() described the 'adjustment' parameter as a dB value. In reality the parameter is a linear gain factor applied directly to the audio samples: a positive value multiplies each sample, a negative value divides each sample, and 0 leaves the audio unchanged. The documentation block has also been reformatted to use the standard ' *' Doxygen style (asterisk in the second column) for consistency with the rest of the file. To convert a dB change to this factor use adjustment = 10^(dB/20) (for example, 2 ~= +6 dB and 10 ~= +20 dB). Resolves: #1167	4 weeks ago
Gian Diego Javes	e21dfae148	taskpool: count the in-flight task when selecting an executor ... The taskpool selectors measure load with ast_taskprocessor_size(), which only reports queued tasks. A taskprocessor that is executing a long running task has an empty queue, so it looks identical to an idle one and the least full selector hands it more work while other executors sit free. Add ast_taskprocessor_is_executing() and include the in-flight task in the load the selectors compare. This is not a new semantic. taskprocessor_push() already counts the in-flight task as outstanding work when it notifies the listener: /* The currently executing task counts as still in queue */ was_empty = tps->executing ? 0 : previous_size == 0; The taskpool selectors are the one place that still measures only the queue. This change applies the same rule there. The accessor reads tps->executing without the object lock, the way the CLI already does in main/taskprocessor.c, because taking the lock on every push cost roughly six times the throughput: taskpool push efficiency went from 324438 to 52966 tasks per second and the serializer variant from 217467 to 38313. Without the lock both are back to baseline. A stale read only means one push is routed as if the taskprocessor had just changed state, which the selector already tolerates. Also add a unit test that holds one executor of a four executor pool inside a long task and checks that a task pushed to a different serializer runs on one of the three free executors. It fails on every run without this change. Fixes: #2074 UserNote: Taskpool executors that are running a long task are no longer considered idle when work is distributed, so tasks are routed to free executors instead of queueing behind a busy one.	4 weeks ago
Mike Bradeen	cc7f16d578	res_pjsip_sdp_rtp: Fix intermittent call drop on reINVITE sdp change ... A reINVITE that changed the offered codec set (e.g. ulaw+alaw -> alaw only) could lead to call teardown when a bridge write still using the old format occurred during the re-negotiation. This change avoids the termination by making the following changes. First, Asterisk now holds the lock across both the tx payload update and the channel format update so they commit atomically. Second, when processing the sdp on the new incoming offer, merge the new offer with the old until the negotiation is complete. Also adds unit tests for the sdp merge Fixes: #2059	2 months ago
Joshua C. Colp	e5b7977c82	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
George Joseph	47388ca3f0	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
Ben Ford	2c90ffaf9d	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	2 months ago
Joshua C. Colp	fdf4e0dc7c	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.	2 months ago
Mehrdad Seifzadeh	6e745a887e	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	2 months ago
Mike Bradeen	08e03e6437	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	2 months ago
George Joseph	83ba36f1ad	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	2 months ago
Joshua C. Colp	2cf19a5cc6	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.	2 months ago
Alexis Chenard	9de4752040	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'.	2 months ago
George Joseph	ec36d7db08	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.	2 months ago
George Joseph	caa4d50d9c	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	4 months ago
Mike Bradeen	e00a99f912	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
George Joseph	6a3ea2540f	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	4 months ago
George Joseph	46a2596a87	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	4 months ago
Peter Krall	68a36c822c	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
George Joseph	d83d8a89fb	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
mattia	88919ce057	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.	4 months ago
George Joseph	ef9d0ccf5d	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	4 months ago
Daniel Donoghue	8f4360f082	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.	4 months ago
Daniel Donoghue	d80869c4e4	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.	4 months ago
Alexei Gradinari	b7dfe75744	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	4 months ago
George Joseph	790f668057	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	7 months ago
Naveen Albert	14ddbf022a	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	7 months ago
George Joseph	d0a30ae399	docs: Add "Provided-by" to doc XML and CLI output. ... For application, function, manager, managerEvent, managerEventInstance and info XML documentation nodes, the make_xml_documentation script will add a "module" attribute if not already present. For XML in separate "*_doc.xml" files, the script figures out the correct module name. For documentation in the "main" directory, the module name is set to "builtin". The CLI handlers for "core show application", "core show function", "manager show command" and "manager show event", have been updated to show the following after the Synopsis... ``` [Provided By] <modulename> ``` For modules that provide additional "info" elements (like the technologies do for Dial), the providing module has also been added. ``` Technology: WebSocket Provided by: chan_websocket WebSocket Dial Strings: ... ``` UserNote: The CLI help for applications, functions, manager commands and manager events now shows the module that provides its functionality.	7 months ago
George Joseph	98ac114678	CDR/CEL Custom Performance Improvements ... There is a LOT of work in this commit but the TL;DR is that it takes CEL processing from using 38% of the CPU instructions used by a call, which is more than that used by the call processing itself, down to less than 10% of the instructions. So here's the deal... cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom all shared one ugly trait...they all used ast_str_substitute_variables() or pbx_substitute_variables_helper() to resolve the dialplan functions used in their config files. Not only are they both extraordinarily expensive, they both require a dummy channel to be allocated and destroyed for each record written. For CDRs, that's not too bad because we only write one CDR per call. For CELs however, it's a disaster. As far as source code goes, the modules basically all did the same thing. Unfortunately, they did it badly. The config files simply contained long opaque strings which were intepreted by ast_str_substitute_variables() or pbx_substitute_variables_helper(), the very functions that ate all the instructions. This meant introducing a new "advanced" config format much like the advanced manager event filtering added to manager.conf in 2024. Fortunately however, if the legacy config was recognizable, we were able to parse it as an advanced config and gain the benefit. If not, then it goes the legacy, and very expensive, route. Given the commonality among the modules, instead of making the same improvements to 4 modules then trying to maintain them over time, a single module "res_cdrel_custom" was created that contains all of the common code. A few bonuses became possible in the process... * The cdr_custom and cel_custom modules now support JSON formatted output. * The cdr_sqlite_custom and cel_sqlite3_custom modules no longer have to share an Sqlite3 database. Summary of changes: A new module "res/res_cdrel_custom.c" has been created and the existing cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules are now just stubs that call the code in res_cdrel_custom. res_cdrel_custom contains: * A common configuration facility. * Getters for both CDR and CEL fields that share the same abstraction. * Formatters for all data types found in the ast_cdr and ast_event structures that share the same abstraction. * Common writers for the text file and database backends that, you guessed it, share the same abstraction. The result is that while there is certainly a net increase in the number of lines in the code base, most of it is in the configuration handling at load-time. The run-time instruction path length is now significanty shorter. ``` Scenario Instructions Latency ===================================================== CEL pre changes 38.49% 37.51% CEL Advanced 9.68% 6.06% CEL Legacy (auto-conv to adv) 9.95% 6.13% CEL Sqlite3 pre changes 39.41% 39.90% CEL Sqlite3 Advanced 25.68% 24.24% CEL Sqlite3 Legacy (auto conv) 25.88% 24.53% CDR pre changes 4.79% 2.95% CDR Advanced 0.79% 0.47% CDR Legacy (auto conv to adv) 0.86% 0.51% CDR Sqlite3 pre changes 4.47% 2.89% CEL Sqlite3 Advanced 2.16% 1.29% CEL Sqlite3 Legacy (auto conv) 2.19% 1.30% ``` Notes: * We only write one CDR per call but every little bit helps. * Sqlite3 still takes a fair amount of resources but the new config makes a decent improvement. * Legacy configs that we can't auto convert will still take the "pre changes" path. If you're interested in more implementation details, see the comments at the top of the res_cdrel_custom.c file. One minor fix to CEL is also included...Although TenantID was added to the ast_event structure, it was always rendered as an empty string. It's now properly rendered. UserNote: Significant performance improvements have been made to the cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules. See the new sample config files for those modules to see how to benefit from them.	7 months ago
Prashant Srivastav	5938c820a1	fix: Add macOS (Darwin) compatibility for building Asterisk ... - Makefile: Skip /usr/lib/bundle1.o on modern macOS (doesn't exist) - Makefile.rules: Skip -fno-partial-inlining for clang (gcc-only flag) - include/asterisk/utils.h: Use asterisk/endian.h instead of <endian.h> - main/Makefile: Add Darwin-specific pjproject linking with -force_load - main/strcompat.c: Include poll-compat.h, use ast_poll() - main/xml.c: Add ASTMM_LIBC ASTMM_IGNORE for libxml2 compatibility - res/res_pjsip/config_transport.c: Define TCP keepalive constants for macOS Tested on macOS Darwin 25.2.0 (Apple Silicon ARM64)	7 months ago
Joshua C. Colp	ab04bf3b39	endpoints: Allow access to latest snapshot directly. ... This change adds an API call to allow direct access to the latest snapshot of an ast_endpoint. This is then used by chan_pjsip when calculating device state, eliminating the need to access the cache which would incur a container find and access.	7 months ago
Naveen Albert	c3db651b23	app_dial, utils.h: Avoid old style declaration and discarded qualifier. ... * app_dial: Use const char* for fixed strings. * utils.h: inline should come before return type. Resolves: #1755	7 months ago
Sean Bright	0e6bdea103	cli.c: Allow 'channel request hangup' to accept patterns. ... This extends 'channel request hangup' to accept multiple channel names, a POSIX Extended Regular Expression, a glob-like pattern, or a combination of all of them. UserNote: The 'channel request hangup' CLI command now accepts multiple channel names, POSIX Extended Regular Expressions, glob-like patterns, or a combination of all of them. See the CLI command 'core show help channel request hangup' for full details.	9 months ago
George Joseph	10060d8dde	chan_websocket: Use the channel's ability to poll fds for the websocket read. ... We now add the websocket's file descriptor to the channel's fd array and let it poll for data availability instead if having a dedicated thread that does the polling. This eliminates the thread and allows removal of most explicit locking since the core channel code will lock the channel to prevent simultaneous calls to webchan_read, webchan_hangup, etc. While we were here, the hangup code was refactored to use ast_hangup_with_cause instead of directly queueing an AST_CONTROL_HANGUP frame. This allows us to set hangup causes and generate snapshots. For a bit of extra debugging, a table of websocket close codes was added to http_websocket.h with an accompanying "to string" function added to res_http_websocket.c Resolves: #1683	9 months ago
George Joseph	8ca3a4978d	res_geolocation: Fix multiple issues with XML generation. ... * 3d positions were being rendered without an enclosing `<gml:pos>` element resulting in invalid XML. * There was no way to set the `id` attribute on the enclosing `tuple`, `device` and `person` elements. * There was no way to set the value of the `deviceID` element. * Parsing of degree and radian UOMs was broken resulting in them appearing outside an XML element. * The UOM schemas for degrees and radians were reversed. * The Ellipsoid shape was missing and the Ellipse shape was defined multiple times. * The `crs` location_info parameter, although documented, didn't work. * The `pos3d` location_info parameter appears in some documentation but wasn't being parsed correctly. * The retransmission-allowed and retention-expiry sub-elements of usage-rules were using the `gp` namespace instead of the `gbp` namespace. In addition to fixing the above, several other code refactorings were performed and the unit test enhanced to include a round trip XML -> eprofile -> XML validation. Resolves: #1667 UserNote: Geolocation: Two new optional profile parameters have been added. * `pidf_element_id` which sets the value of the `id` attribute on the top-level PIDF-LO `device`, `person` or `tuple` elements. * `device_id` which sets the content of the `<deviceID>` element. Both parameters can include channel variables. UpgradeNote: Geolocation: In order to correct bugs in both code and documentation, the following changes to the parameters for GML geolocation locations are now in effect: * The documented but unimplemented `crs` (coordinate reference system) element has been added to the location_info parameter that indicates whether the `2d` or `3d` reference system is to be used. If the crs isn't valid for the shape specified, an error will be generated. The default depends on the shape specified. * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs. If crs isn't specified, it will default to `2d` for these shapes. The Sphere, Ellipsoid and Prism shapes MUST use a `3d` crs. If crs isn't specified, it will default to `3d` for these shapes. The Point and Polygon shapes may use either crs. The default crs is `2d` however so if `3d` positions are used, the crs must be explicitly set to `3d`. * The `geoloc show gml_shape_defs` CLI command has been updated to show which coordinate reference systems are valid for each shape. * The `pos3d` element has been removed in favor of allowing the `pos` element to include altitude if the crs is `3d`. The number of values in the `pos` element MUST be 2 if the crs is `2d` and 3 if the crs is `3d`. An error will be generated for any other combination. * The angle unit-of-measure for shapes that use angles should now be included in the respective parameter. The default is `degrees`. There were some inconsistent references to `orientation_uom` in some documentation but that parameter never worked and is now removed. See examples below. Examples... ``` location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20" location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620" location_info = shape="Point", pos="39.0 -105.0" location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20" semiMinorAxis="10", verticalAxis="0", orientation="25 degrees" pidf_element_id = ${CHANNEL(name)}-${EXTEN} device_id = mac:001122334455 Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN}) ```	9 months ago
Maximilian Fridrich	3fa6e90d2d	res_pjsip: Introduce redirect module for handling 3xx responses ... This commit introduces a new redirect handling module that provides infrastructure for following SIP 3xx redirect responses. The redirect functionality respects the endpoint's redirect_method setting and only follows redirects when set to 'uri_pjsip'. This infrastructure can be used by any PJSIP module that needs to handle 3xx redirect responses.	9 months ago
Justin T. Gibbs	bd8a510a6b	rtp/rtcp: Configure dual-stack behavior via IPV6_V6ONLY ... Dual-stack behavior (simultaneous listening for IPV4 and IPV6 connections on a single socket) is required by Asterisk's ICE implementation. On systems with the IPV6_V6ONLY sockopt, set the option to 0 (dual-stack enabled) when binding to the IPV6 any address. This ensures correct behavior regardless of the system's default dual-stack configuration.	9 months ago
Joshua C. Colp	44a68fb442	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	9 months ago
Sean Bright	45ffbb3f3a	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	9 months ago
Mike Bradeen	b42bf1d319	taskprocessors: Improve logging and add new cli options ... This change makes some small changes to improve log readability in addition to the following changes: Modified 'core show taskprocessors' to now show Low time and High time for task execution. New command 'core show taskprocessor name <taskprocessor-name>' to dump taskprocessor info and current queue. Addionally, a new test was added to demonstrate the 'show taskprocessor name' functionality: test execute category /main/taskprocessor/ name taskprocessor_cli_show Setting 'core set debug 3 taskprocessor.c' will now log pushed tasks. (Warning this is will cause extremely high levels of logging at even low traffic levels.) Resolves: #1566 UserNote: New CLI command has been added - core show taskprocessor name <taskprocessor-name>	9 months ago
George Joseph	7c9133c854	ccss: Add option to ccss.conf to globally disable it. ... The Call Completion Supplementary Service feature is rarely used but many of it's functions are called by app_dial and channel.c "just in case". These functions lock and unlock the channel just to see if CCSS is enabled on it, which it isn't 99.99% of the time. UserNote: A new "enabled" parameter has been added to ccss.conf. It defaults to "yes" to preserve backwards compatibility but CCSS is rarely used so setting "enabled = no" in the "general" section can save some unneeded channel locking operations and log message spam. Disabling ccss will also prevent the func_callcompletion and chan_dahdi modules from loading. DeveloperNote: A new API ast_is_cc_enabled() has been added. It should be used to ensure that CCSS is enabled before making any other ast_cc_* calls.	9 months ago
Joshua C. Colp	54da3e1b9e	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	12 months ago
Joshua C. Colp	7662c5cb6c	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	12 months ago
Christoph Moench-Tegeder	9bbb566df5	Fix Endianness detection in utils.h for non-Linux ... Commit 43bf8a4ded introduced endian dependend byte-swapping code in include/asterisk/utils.h, where the endianness was detected using the __BYTE_ORDER macro. This macro lives in endian.h, which on Linux is included implicitely (by the network-related headers, I think), but on FreeBSD the headers are laid out differently and we do not get __BYTE_ORDER the implicit way. Instead, this makes the usage of endian.h explicit by including it where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the POSIX-defined ones (see https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html for standard compliance). Additionally, this adds a compile-time check for the endianness-logic: compilation will fail if neither big nor little endian can be detected. Fixes: #1536	12 months ago
George Joseph	1b98cd1b25	chan_pjsip: Add technology-specific off-nominal hangup cause to events. ... Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI and AMI hangup and channel destroyed events, it can be helpful to know what the actual channel technology code was if the call was unsuccessful. For PJSIP, it's the SIP response code. * A new "tech_hangupcause" field was added to the ast_channel structure along with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set() functions. It should only be set for off-nominal terminations. * chan_pjsip was modified to set the tech hangup cause in the chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit tricky because these two functions aren't always called in the same order. The channel that hangs up first will get chan_pjsip_session_end() called first which will trigger the core to call chan_pjsip_hangup() on itself, then call chan_pjsip_hangup() on the other channel. The other channel's chan_pjsip_session_end() function will get called last. Unfortunately, the other channel's HangupRequest events are sent before chan_pjsip has had a chance to set the tech hangupcause code so the HangupRequest events for that channel won't have the cause code set. The ChannelDestroyed and Hangup events however will have the code set for both channels. * A new "tech_cause" field was added to the ast_channel_snapshot_hangup structure. This is a public structure so a bit of refactoring was needed to preserve ABI compatibility. * The ARI ChannelHangupRequest and ChannelDestroyed events were modified to include the "tech_cause" parameter in the JSON for off-nominal terminations. The parameter is suppressed for nominal termination. * The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to include the "TechCause" parameter for off-nominal terminations. Like their ARI counterparts, the parameter is suppressed for nominal termination. DeveloperNote: A "tech_cause" parameter has been added to the ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" parameter has been added to the HangupRequest, SoftHangupRequest and Hangup AMI event messages. For chan_pjsip, these will be set to the last SIP response status code for off-nominally terminated calls. The parameter is suppressed for nominal termination.	12 months ago
Sven Kube	da10edda02	res_audiosocket: add message types for all slin sample rates ... Extend audiosocket messages with types 0x11 - 0x18 to create asterisk frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and slin192 format, enabling the transmission of audio at a higher sample rates. For audiosocket messages sent by Asterisk, the message kind is determined by the format of the originating asterisk frame. UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and slin192 audio. External applications using audiosocket may need to be updated to support these message types if the audiosocket channel is created with one of these audio formats.	12 months ago
Joshua C. Colp	5b4aac6ba0	endpoints: Remove need for stasis subscription. ... When an endpoint is created in the core of Asterisk a subscription was previously created alongside it to monitor any channels being destroyed that were related to it. This was done by receiving all channel snapshot updates for every channel and only reacting when it was indicated that the channel was dead. This change removes this logic and instead provides an API call for directly removing a channel from an endpoint. This is called when channels are destroyed. This operation is fast, so blocking the calling thread for a short period of time doesn't have any noticeable impact.	12 months ago