7047 Commits (master)

Author	SHA1	Message	Date
phoneben	9a7fd6d816	main/lock: log failed mutex unlock in non-DEBUG_THREADS builds ... main/lock: log failed mutex unlock in non-DEBUG_THREADS builds In non-DEBUG_THREADS builds, a failed pthread_mutex_unlock() is silent, so an unlock by a non-owning thread leaves the mutex locked and the only symptom is a later hang. Log an error naming the mutex, file, line and function when the unlock fails. The success path costs one branch. Tested on 20.21.0 with a throwaway module covering both a never-locked unlock and a cross-thread unlock. AI assistance: Claude (Anthropic) helped draft this. I reviewed and tested the change. Resolves: #2233	2 days ago
phoneben	6b88b7d0b2	file.c: Log the actual file extension during playback ... file.c: Log the actual file extension during playback The playback verbose message currently uses the channel write format when building the displayed filename. This can be misleading. For example, a WAV file may be logged as: Playing 'soundfile.slin' because WAV playback uses signed linear audio internally. Store the actual file extension matched by filehelper() on the ast_filestream and use that extension in the playback verbose message. This allows the log to reflect the actual source file that was opened, including cases where the format handler name differs from the file extension. Example after the change: Playing 'soundfile.wav' This only changes the verbose log output and does not affect playback behavior. AI assistance: ChatGPT assisted with identifying the issue and PR text.	2 weeks ago
Evgeniy Kravtsov	ee51d78c68	cdr: Preserve ANSWERED disposition with ignoredialchanges ... When ignoredialchanges is enabled, Dial End events update the disposition of the single CDR. With calls that dial multiple destinations, such as Queue(), one destination can answer while other dialed channels are subsequently cancelled. Those later Dial End events currently overwrite ANSWERED with a non-answer disposition even though the call was answered. Once the CDR reaches AST_CDR_ANSWERED, retain that disposition while dial changes are ignored. Fixes: #2181 UserNote: With ignoredialchanges enabled, answered calls retain an ANSWERED disposition when later dial attempts end without answer.	3 weeks ago
Sean Bright	b62d340366	hashtab.c: Object lookups can be const ... The objects passed to the various hashtable removal functions are not modified, so they can - and should - be `const`.	3 weeks ago
Sean Bright	5841ae7248	file.c: Remove long-deprecated `languageprefix` option ... This was deprecated in afbfafa4 on 2006-12-27. UpgradeNote: The `languageprefix` configuration parameter in `asterisk.conf`, deprecated in 2006, as been removed. The sounds path will now always include the language prefix.	3 weeks ago
Mike Bradeen	c097df5d41	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.	3 weeks ago
Joshua C. Colp	5750f7c608	extstate/pjsip: Handle lack of or late loading of presence state. ... The extension state implementation is stateful resulting in state not being updated if a presence state provider does not publish an update after loading. To handle providers which don't do this we query for updated state if we are asked for it and the existing state is invalid. This ensures that if the presence state provider is loaded later we will get the updated state when asked. The func_presencestate module is written to publish the persisted custom presence states at load time. This failed for entries with a not_set state as this resulted in querying the presence provider that was not yet registered. The provider is now registered before this ensuring the update can be published. This change also fixes an issue in PJSIP where a SUBSCRIBE for a hint that had an invalid presence state configuration would not send an initial NOTIFY. We now treat an invalid presence state as not set instead. Fixes: #2179	3 weeks ago
Joshua C. Colp	e73cceb5b7	extstate: Include causing device channel details in legacy callbacks. ... When a legacy extension state callback registers it can optionally do so in a way that it will receive the channel(s) that have caused a device to ring. This is used in PJSIP for the dialog-info+xml support so that phones can display who is calling the monitored extension. An issue was introduced in the extension state rewrite where this did not occur for the actual device that caused the extension state update. It was only being done for additional devices that could be involved. This has been fixed so that it is now done for the causing device. Fixes: #2186	3 weeks ago
Sean Bright	291d170b73	say.c: Look for "and" in several locations, falling back to `vm-and` ... `vm-and` exists in all of the core sounds files that the project distributes, so while we will now try to find `digits/and` and `and` first, we always fallback to `vm-and`. Resolves: #622	3 weeks ago
Sean Bright	b768dec4a4	file.c: Reduce number of allocations when searching for sound files ... Depending on the requested format, the number of format modules loaded, and the order those modules were loaded, it's possible for dozens of small allocations to occur for each file playback on a channel. So instead we now allocate a single buffer big enough for any registered file format extension and we use that for all lookups.	1 month ago
Jeremy Lainé	ffbc0d6bab	main/file.c: Log the reason a file could not be opened. ... When ast_streamfile() failed to open a file it reported strerror(errno), but errno was not set by anything on that path. The value came from whatever syscall ran last, so the message routinely blamed an unrelated error. Drop errno from that message and log at the point of failure instead: filehelper() now reports the fopen() error, which does carry a meaningful errno, and an allocation failure from get_filestream(). Candidate files whose format does not match the channel are logged at debug level only, since filehelper() walks every registered format and skipping the ones that do not match is normal: a sound installed in several formats would otherwise warn on every successful playback. Failures inside the format module's open callback stay unlogged here, as those callbacks already report their own reason. Resolves: #1629	2 months ago
Gian Diego Javes	6d383d321d	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.	2 months ago
Mike Bradeen	f0c1c1a7b7	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	3 months ago
Naveen Albert	f6b02c640c	backtrace.c: Include stdbool.h ... stdbool.h is needed now that we use the bool type directly instead of bfd_boolean. See also 1d95b744c06974f0a00c143c6e0fc979af930908. Resolves: #2061	3 months ago
Naveen Albert	5b59829015	backtrace.c: Avoid removed bfd_boolean type. ... bfd_boolean has been deprecated for some time and has now been removed in gdb, so use bool directly instead. See: 1d95b744c0 Resolves: #2054	3 months ago
Jeremy Lainé	7c60726df1	tcptls: Don't inhibit escalations for outbound client connections. ... handle_tcptls_connection() marks the current thread as inhibiting privilege escalations and as an external user interface so that dialplan functions considered 'dangerous' (STAT, SHELL, ...) cannot be executed on behalf of an external protocol. This is correct for inbound (server) connections, each of which runs on its own dedicated worker thread. However, the outbound (client) path calls handle_tcptls_connection() synchronously on the caller's own thread. When that caller is a channel/PBX thread, the thread-local flags are set and never cleared, permanently tainting the dialplan thread. Examples: - Calling Dial() for an outbound WebSocket and resuming dialplan execution with the "g" flag. - Running ExternalIVR() then continuing dialplan executing. Outbound connections are initiated by Asterisk itself and are not external user interfaces, so the flags should not be set for them. Gate the flag-setting on tcptls_session->client so it applies only to inbound connections. Resolves: #2038	3 months ago
Jeremy Lainé	b31d3b1d57	file.c: Ensure opening a stream opens at most one file ... The functions used to stream files eventually end up calling into `filehelper(.., ACTION_OPEN)` which takes care of iterating over supported extensions and opening the first existing file. To do this, `filehelper` uses two nested loops: - An outer loop over the supported file formats. - An inner loop over the possible extensions for that format. We need to break out of both loops as soon as a file was successfully opened. Otherwise if a file exists in multiple formats (e.g `foo.wav` and `foo.alaw`) both files will be opened successively. Resolves: #1625	3 months ago
Roel van Meer	e22687d0bd	bridging: Add support for TOUCH_MIXMONITOR_OPTIONS ... This channel variable can be used to override the default automixmon options (which are 'b' or 'bB(<n>)'). If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning. It is the responsibility of the user to configure valid options in TOUCH_MIXMONITOR_OPTIONS. Fixes: #1981 UserNote: The TOUCH_MIXMONITOR_OPTIONS channel variable can now be used to configure the options used by the mixmonitor application when it is started via automixmon. If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning.	3 months ago
George Joseph	1480af1227	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	3 months ago
Mike Bradeen	fb4bfb1156	Audiohooks: Allow whisper audiohooks to work without an underlying stream ... Audiohook whispers work by mixing audio with the current stream of frames being written to the hooked channel. Prior to this change if there were no audio frames being written to the channel outside of the audiohook, then any whisper audio has nothing to mix with and so is not heard on the hooked channel. This change removes that requirement by creating a framehook driven timer on the channel that creates silent frames the whispered frames can be mixed into. The framehook is automatically added with the first attached whisper audiohook and is removed when the last one is detached. If there are no attached streams being written to the channel, the timer is used to generate a default one. If a stream is attached; the timer is ignored until the stream stops. This stop and start is automatically triggered by the presence of a written stream and does not require any external action. Resolves: #1966 UserNote: There is no longer a requirement to play silence or otherwise have a stream of audio being sent to a hooked channel for whispered frames to be heard.	3 months ago
Ben Ford	99fe892b66	Logger: Don't discard WARNING or ERROR messages. ... When the logger reaches it's threshold for messages, don't discard the WARNING or ERROR messages up to a certain amount. If THAT threshold reaches its maximum, start discarding the oldest non-WARNING/ERROR messages until we only have WARNING and ERROR messages left. Everything will be discarded after that until we have room again for more. Also added a test that limits the queue sizes and logs many messages to see the queues hitting their maximums and discarding the appropriate messages. Fixes: #1882	3 months ago
Joshua C. Colp	2158dff4b5	manager: Move away from shared linked list for events. ... This change moves manager away from a shared linked list for events to a per-session vector of pending events. This allows early filtering of events which eliminates the need to wake up threads for events that they have no interest in. TCP based connections have also been moved to an alert pipe based wakeup mechanism to reduce locking and simplify usage. UserNote: The "manager show eventq" CLI command has been removed as there is no longer a single event queue to display.	4 months ago
Alexandre Fournier	a9d33edc2e	format_cap: guard against NULL src in *_from_cap helpers ... ast_format_cap_append_from_cap() and ast_format_cap_replace_from_cap() dereference 'src' (src->preference_order) without checking it for NULL. A dummy channel allocated with ast_dummy_channel_alloc() never sets a native-format capability, so ast_channel_nativeformats() returns NULL on such channels. When CHANNEL(audionativeformat) / CHANNEL(videonativeformat) is evaluated against a dummy channel (e.g. via ARI channelvars during a Stasis VarSet event raised while app_voicemail builds the notification email on a dummy channel), func_channel_read() passes that NULL straight into ast_format_cap_append_from_cap(), causing a NULL dereference at offset 0x28 and a SIGSEGV. Guard both helpers against a NULL source. A NULL source simply means "no formats to copy", so appending/replacing nothing is the correct no-op behaviour. This also protects all other callers. Fixes https://github.com/asterisk/asterisk/issues/1992 AI disclosure: this was generated using Claude Opus 4.8, tested to fix the issue. Not sure if it is the *right* way to do it.	4 months ago
Mike Bradeen	87a1f43e4c	manager: Use remote address in user error logging ... To avoid a potential null dereference use the remote address in error logging when there is no user or the user acl fails. Resolves: #GHSA-3rhj-hhw7-m6fw	4 months ago
Milan Kyselica	4b360a61d6	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
George Joseph	f9a38e1608	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	4 months ago
Joshua C. Colp	f4f6ec0313	extension_state: Add new extension state API. ... Extension state to this point has been an API implemented inside the PBX core resulting in its state being intermingled with that of the dialplan. This increased the complexity of the PBX core and made it difficult to enact improvements. This change adds a new separate extension state API which receives updates from the PBX core as configuration changes but maintains its own separate state. The API is also written to fully take advantage of modern APIs in a more selective manner by subscribing each extension state to only the devices it is interested in, ultimately reducing resource consumption during updates. Presence state updates being infrequently done use a single shared subscription that goes through the extension states to find and update ones that the update is applicable to. Legacy API support is provided by reimplementing the API as wrappers over the new extension state API. This improves the legacy API by making it multithreaded, with each callback being individually subscribed. Autohints support is maintained but has been separated out into a self contained new implementation. Synchronous subscription support has also been added to Stasis to remove the overhead of asynchronous publishing when the handling of published messages is small and fast.	4 months ago
George Joseph	184fafc20c	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.	5 months ago
Mike Bradeen	19eb1897da	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
Bernd Kuhls	4409c49604	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
Alexander Bakker	3a9e3c3623	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
George Joseph	0515585bda	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
George Joseph	22f0a53ce4	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	1146e1943d	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
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.	6 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	6 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.	6 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	6 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	6 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	6 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	7 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	7 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	7 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".	7 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	7 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	7 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	8 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	8 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	8 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	8 months ago