4430 Commits (3009c8cc7e4b9bee28d3f8da42660fbc310e5214)

Author	SHA1	Message	Date
Joshua C. Colp	fed0d484a5	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	3 months ago
Joshua C. Colp	da46b5f33f	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.	3 months ago
Christoph Moench-Tegeder	bced61f2be	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	3 months ago
George Joseph	07caf2ccda	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.	3 months ago
Sven Kube	14d82ecf62	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.	3 months ago
Joshua C. Colp	5dfb2f7200	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.	3 months ago
George Joseph	411f66bd18	taskpool: Fix some references to threadpool that should be taskpool. ... Resolves: #1478	3 months ago
Igor Goncharovsky	9ee4d452db	func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() ... As soon as SIP call may end with several Reason headers, we want to make all of them available through the HAGUPCAUSE() function. This implementation uses the same ao2 hash for cause codes storage and adds a flag to make difference between last processed sip message and content of reason headers. UserNote: Added a new option to HANGUPCAUSE to access additional information about hangup reason. Reason headers from pjsip could be read using 'tech_extended' cause type.	3 months ago
Bastian Triller	c2e03d8e8e	Fix some doxygen, typos and whitespace	3 months ago
Joshua C. Colp	061cc9562a	taskpool: Add taskpool API, switch Stasis to using it. ... This change introduces a new API called taskpool. This is a pool of taskprocessors. It provides the following functionality: 1. Task pushing to a pool of taskprocessors 2. Synchronous tasks 3. Serializers for execution ordering of tasks 4. Growing/shrinking of number of taskprocessors in pool This functionality already exists through the combination of threadpool+taskprocessors but through investigating I determined that this carries substantial overhead for short to medium duration tasks. The threadpool uses a single queue of work, and for management of threads it involves additional tasks. I wrote taskpool to eliminate the extra overhead and management as much as possible. Instead of a single queue of work each taskprocessor has its own queue and at push time a selector chooses the taskprocessor to queue the task to. Each taskprocessor also has its own thread like normal. This spreads out the tasks immediately and reduces contention on shared resources. Using the included efficiency tests the number of tasks that can be executed per second in a taskpool is 6-12 times more than an equivalent threadpool+taskprocessor setup. Stasis has been moved over to using this new API as it is a heavy consumer of threadpool+taskprocessors and produces a lot of tasks. UpgradeNote: The threadpool_* options in stasis.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 stasis. DeveloperNote: The taskpool API has been added for common usage of a pool of taskprocessors. It is suggested to use this API instead of the threadpool+taskprocessor approach.	3 months ago
Alexei Gradinari	dfeb54a00f	sorcery: Prevent duplicate objects and ensure missing objects are created on update ... This patch resolves two issues in Sorcery objectset handling with multiple backends: 1. Prevent duplicate objects: When an object exists in more than one backend (e.g., a contact in both 'astdb' and 'realtime'), the objectset previously returned multiple instances of the same logical object. This caused logic failures in components like the PJSIP registrar, where duplicate contact entries led to overcounting and incorrect deletions, when max_contacts=1 and remove_existing=yes. This patch ensures only one instance of an object with a given key is added to the objectset, avoiding these duplicate-related side effects. 2. Ensure missing objects are created: When using multiple writable backends, a temporary backend failure can lead to objects missing permanently from that backend. Currently, .update() silently fails if the object is not present, and no .create() is attempted. This results in inconsistent state across backends (e.g. astdb vs. realtime). This patch introduces a new global option in sorcery.conf: [general] update_or_create_on_update_miss = yes|no Default: no (preserves existing behavior). When enabled: if .update() fails with no data found, .create() is attempted in that backend. This ensures that objects missing due to temporary backend outages are re-synchronized once the backend is available again. Added a new CLI command: sorcery show settings Displays global Sorcery settings, including the current value of update_or_create_on_update_miss. Updated tests to validate both flag enabled/disabled behavior. Fixes: #1289 UserNote: Users relying on Sorcery multiple writable backends configurations (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes in sorcery.conf to ensure missing objects are recreated after temporary backend failures. Default behavior remains unchanged unless explicitly enabled. (cherry picked from commit a6142af2a0)	5 months ago
George Joseph	5d1d5dbfea	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. ... * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `core show application Dial` that shows how to use it. Resolves: #1352 UserNote: A new WebSocket channel driver option `v` has been added to the Dial application that allows you to specify additional URI parameters on outgoing connections. Run `core show application Dial` from the Asterisk CLI to see how to use it. (cherry picked from commit de6aaa9623)	5 months ago
Sven Kube	996db4c8d8	ARI: Add command to indicate progress to a channel ... Adds an ARI command to send a progress indication to a channel. DeveloperNote: A new ARI endpoint is available at `/channels/{channelId}/progress` to indicate progress to a channel. (cherry picked from commit a0b36f8099)	5 months ago
Sperl Viktor	bb801f47c1	cel: Add STREAM_BEGIN, STREAM_END and DTMF event types. ... Fixes: #1280 UserNote: Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types in cel.conf will log media files and music on hold played to each channel. The STREAM_BEGIN event's extra field will contain a JSON with the file details (path, format and language), or the class name, in case of music on hold is played. The DTMF event's extra field will contain a JSON with the digit and the duration in milliseconds. (cherry picked from commit 0906282f25)	5 months ago
zhou_jiajian	89d3c2e92b	cdr: add CANCEL dispostion in CDR ... In the original implementation, both CANCEL and NO ANSWER states were consolidated under the NO ANSWER disposition. This patch introduces a separate CANCEL disposition, with an optional configuration switch to enable this new disposition. Resolves: #1323 UserNote: A new CDR option "canceldispositionenabled" has been added that when set to true, the NO ANSWER disposition will be split into two dispositions: CANCEL and NO ANSWER. The default value is 'no' (cherry picked from commit 9d4674d93d)	5 months ago
George Joseph	a04c10aec2	options: Change ast_options from ast_flags to ast_flags64. ... DeveloperNote: The 32-bit ast_options has no room left to accomodate new options and so has been converted to an ast_flags64 structure. All internal references to ast_options have been updated to use the 64-bit flag manipulation macros. External module references to the 32-bit ast_options should continue to work on little-endian systems because the least-significant bytes of a 64 bit integer will be in the same location as a 32-bit integer. Because that's not the case on big-endian systems, we've swapped the bytes in the flags manupulation macros on big-endian systems so external modules should still work however you are encouraged to test. (cherry picked from commit d1a1945967)	5 months ago
Alexei Gradinari	3a2cedace0	res_config_odbc: Prevent Realtime fallback on record-not-found (SQL_NO_DATA) ... This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly falls back to the next configured backend when the current one returns SQL_NO_DATA (i.e., no record found). This is a logical error and performance risk in multi-backend configurations. Solution: Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker. ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found. Core engine stops iterating on this marker, avoiding unnecessary fallback. Notes: Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly. This patch only covers ODBC. Fixes: #1305 (cherry picked from commit 9e3ea1c8ac)	5 months ago
George Joseph	df33c52a1f	Media over Websocket Channel Driver ... * Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information. * res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs. * main/channel.c: Changed an obsolete nuisance error to a debug. * ARI channels: Updated externalMedia to include chan_websocket as a supported transport. UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket. (cherry picked from commit 07fd3af897)	5 months ago
mkmer	9232cde023	utils.h: Add rounding to float conversion to int. ... Quote from an audio engineer NR9V: There is a minor issue of a small amount of crossover distortion though as a result of `ast_slinear_saturated_multiply_float()` not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion. Resolves: #1176 (cherry picked from commit 55f4294e82)	5 months ago
Tinet-mucw	3b9d5a00e4	pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return -1. ... Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by GotoIf this info is changed. that will causes confusion in dialplan execution. Resolves: #1273 (cherry picked from commit 88f653c12d)	5 months ago
George Joseph	f82c7b2694	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	5 months ago
George Joseph	04a3e854d0	res_websocket_client: Create common utilities for websocket clients. ... Since multiple Asterisk capabilities now need to create websocket clients it makes sense to create a common set of utilities rather than making each of those capabilities implement their own. * A new configuration file "websocket_client.conf" is used to store common client parameters in named configuration sections. * APIs are provided to list and retrieve ast_websocket_client objects created from the named configurations. * An API is provided that accepts an ast_websocket_client object, connects to the remote server with retries and returns an ast_websocket object. TLS is supported as is basic authentication. * An observer can be registered to receive notification of loaded or reloaded client objects. * An API is provided to compare an existing client object to one just reloaded and return the fields that were changed. The caller can then decide what action to take based on which fields changed. Also as part of thie commit, several sorcery convenience macros were created to make registering common object fields easier. UserNote: A new module "res_websocket_client" and config file "websocket_client.conf" have been added to support several upcoming new capabilities that need common websocket client configuration.	5 months ago
George Joseph	46549fa4e3	asterisk.c: Add option to restrict shell access from remote consoles. ... UserNote: A new asterisk.conf option 'disable_remote_console_shell' has been added that, when set, will prevent remote consoles from executing shell commands using the '!' prefix. Resolves: #GHSA-c7p6-7mvq-8jq2	5 months ago
George Joseph	e7d990b66b	Alternate Channel Storage Backends ... Full details: http://s.asterisk.net/dc679ec3 The previous proof-of-concept showed that the cpp_map_name_id alternate storage backed performed better than all the others so this final PR adds only that option. You still need to enable it in menuselect under the "Alternate Channel Storage Backends" category. To select which one is used at runtime, set the "channel_storage_backend" option in asterisk.conf to one of the values described in asterisk.conf.sample. The default remains "ao2_legacy". UpgradeNote: With this release, you can now select an alternate channel storage backend based on C++ Maps. Using the new backend may increase performance and reduce the chances of deadlocks on heavily loaded systems. For more information, see http://s.asterisk.net/dc679ec3	5 months ago
Naveen Albert	bfbbe28344	sig_analog: Add Call Waiting Deluxe support. ... Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	5 months ago
George Joseph	74f50b12b3	lock.h: Add include for string.h when DEBUG_THREADS is defined. ... When DEBUG_THREADS is defined, lock.h uses strerror(), which is defined in the libc string.h file, to print warning messages. If the including source file doesn't include string.h then strerror() won't be found and and compile errors will be thrown. Since lock.h depends on this, string.h is now included from there if DEBUG_THREADS is defined. This way, including source files don't have to worry about it.	5 months ago
Albrecht Oster	7ce8720528	res_pjproject: Fix DTLS client check failing on some platforms ... Certain platforms (mainly BSD derivatives) have an additional length field in `sockaddr_in6` and `sockaddr_in`. `ast_sockaddr_from_pj_sockaddr()` does not take this field into account when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. The resulting `ast_sockaddr` will have an uninitialized value for `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` has the correct length value set. This has the effect that `ast_sockaddr_cmp()` will always indicate an address mismatch, because it does a bitwise comparison, and all DTLS packets are dropped even if addresses and ports match. `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields are available on the current platform and sets the values accordingly. Resolves: #505 (cherry picked from commit 5f9cd260ac)	9 months ago
George Joseph	a6dca5bf3a	Prequisites for ARI Outbound Websockets ... stasis: * Added stasis_app_is_registered(). * Added stasis_app_control_mark_failed(). * Added stasis_app_control_is_failed(). * Fixed res_stasis_device_state so unsubscribe all works properly. * Modified stasis_app_unregister() to unsubscribe from all event sources. * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() returns true. http: * Added ast_http_create_basic_auth_header(). md5: * Added define for MD5_DIGEST_LENGTH. tcptls: * Added flag to ast_tcptls_session_args to suppress connection log messages to give callers more control over logging. http_websocket: * Add flag to ast_websocket_client_options to suppress connection log messages to give callers more control over logging. * Added username and password to ast_websocket_client_options to support outbound basic authentication. * Added ast_websocket_result_to_str(). (cherry picked from commit cc92adc5fb)	9 months ago
George Joseph	8d69af73d6	ARI: REST over Websocket ... This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ (cherry picked from commit 4e07d3d50f)	9 months ago
mkmer	d6a1b044a8	audiohook.c: Add ability to adjust volume with float ... Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. This is accomplished by the following: Convert internal variables to type float. Always use ast_frame_adjust_volume_float() for adjustments. Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). Cast float to int in ast_audiohook_volume_get() Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. This update maintains 100% backward compatibility. Resolves: #1171 (cherry picked from commit 874fde5924)	9 months ago
Florent CHAUVEAU	d032df902b	audiosocket: added support for DTMF frames ... Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). (cherry picked from commit c44e7e85ec)	9 months ago
Norm Harrison	423d2bad16	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' ... Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit 5cfc79ebea)	9 months ago
Norm Harrison	428c14d1f1	audiosocket: fix timeout, fix dialplan app exit, server address in logs ... - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit 218c64c4eb)	9 months ago
George Joseph	ac843c3600	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. ... Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211 (cherry picked from commit 6cf62b6032)	11 months ago
George Joseph	787ffd5f47	res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. ... The verification check for missing or anonymous callerid was happening before the endpoint's profile was retrieved which meant that the failure_action parameter wasn't available. Therefore, if verification was enabled and there was no callerid or it was "anonymous", the call was immediately terminated instead of giving the dialplan the ability to decide what to do with the call. * The callerid check now happens after the verification context is created and the endpoint's stir_shaken_profile is available. * The check now processes the callerid failure just as it does for other verification failures and respects the failure_action parameter. If set to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)` in the dialplan will return "invalid_or_no_callerid". * If the endpoint's failure_action is "reject_request", the call will be rejected with `433 "Anonymity Disallowed"`. * If the endpoint's failure_action is "continue_return_reason", the call will continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"` header will be added to the next provisional or final response. Resolves: #1112 (cherry picked from commit 0b6a3df331)	11 months ago
Holger Hans Peter Freyther	6f17f61394	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function. (cherry picked from commit 5e4fca062c)	11 months ago
Sean Bright	d402b59ec2	channel.c: Remove dead AST_GENERATOR_FD code. ... Nothing ever sets the `AST_GENERATOR_FD`, so this block of code will never execute. It also is the only place where the `generate` callback is called with the channel lock held which made it difficult to reason about the thread safety of `ast_generator`s. In passing, also note that `AST_AGENT_FD` isn't used either. (cherry picked from commit 0883e5e44d)	11 months ago
George Joseph	c83a6fb64a	res_pjsip: Fix startup/reload memory leak in config_auth. ... An issue in config_auth.c:ast_sip_auth_digest_algorithms_vector_init() was causing double allocations for the two supported_algorithms vectors to the tune of 915 bytes. The leak only happens on startup and when a reload is done and doesn't get bigger with the number of auth objects defined. * Pre-initialized the two vectors in config_auth:auth_alloc(). * Removed the allocations in ast_sip_auth_digest_algorithms_vector_init(). * Added a note to the doc for ast_sip_auth_digest_algorithms_vector_init() noting that the vector passed in should be initialized and empty. * Simplified the create_artificial_auth() function in pjsip_distributor. * Set the vector initialization count to 0 in config_global:global_apply().	1 year ago
Allan Nathanson	3a261ebed4	config.c: fix #tryinclude being converted to #include on rewrite ... Correct an issue in ast_config_text_file_save2() when updating configuration files with "#tryinclude" statements. The API currently replaces "#tryinclude" with "#include". The API also creates empty template files if the referenced files do not exist. This change resolves these problems. Resolves: https://github.com/asterisk/asterisk/issues/920 (cherry picked from commit 4e5df9e12a)	1 year ago
George Joseph	8ca64f1d7f	docs: Enable since/version handling for XML, CLI and ARI documentation ... * Added the "since" element to the XML configObject and configOption elements in appdocsxml.dtd. * Added the "Since" section to the following CLI output: ``` config show help <module> <object> config show help <module> <object> <option> core show application <app> core show function <func> manager show command <command> manager show event <event> agi show commands topic <topic> ``` * Refactored the commands above to output their sections in the same order: Synopsis, Since, Description, Syntax, Arguments, SeeAlso * Refactored the commands above so they all use the same pattern for writing the output to the CLI. * Fixed several memory leaks caused by failure to free temporary output buffers. * Added a "since" array to the mustache template for the top-level resources (Channel, Endpoint, etc.) and to the paths/methods underneath them. These will be added to the generated markdown if present. Example: ``` "resourcePath": "/api-docs/channels.{format}", "requiresModules": [ "res_stasis_answer", "res_stasis_playback", "res_stasis_recording", "res_stasis_snoop" ], "since": [ "18.0.0", "21.0.0" ], "apis": [ { "path": "/channels", "description": "Active channels", "operations": [ { "httpMethod": "GET", "since": [ "18.6.0", "21.8.0" ], "summary": "List all active channels in Asterisk.", "nickname": "list", "responseClass": "List[Channel]" }, ``` NOTE: No versioning information is actually added in this commit. Those will be added separately and instructions for adding and maintaining them will be published on the documentation site at a later date. (cherry picked from commit c010fd4689)	1 year ago
Artem Umerov	24e0a97c67	logger.h: Fix build when AST_DEVMODE is not defined. ... Resolves: #1058 (cherry picked from commit 4c8c37b21b)	1 year ago
George Joseph	10664757d2	Add SHA-256 and SHA-512-256 as authentication digest algorithms ... * Refactored pjproject code to support the new algorithms and added a patch file to third-party/pjproject/patches * Added new parameters to the pjsip auth object: * password_digest = <algorithm>:<digest> * supported_algorithms_uac = List of algorithms to support when acting as a UAC. * supported_algorithms_uas = List of algorithms to support when acting as a UAS. See the auth object in pjsip.conf.sample for detailed info. * Updated both res_pjsip_authenticator_digest.c (for UAS) and res_pjsip_outbound_authentocator_digest.c (UAC) to suport the new algorithms. The new algorithms are only available with the bundled version of pjproject, or an external version > 2.14.1. OpenSSL version 1.1.1 or greater is required to support SHA-512-256. Resolves: #948 UserNote: The SHA-256 and SHA-512-256 algorithms are now available for authentication as both a UAS and a UAC. (cherry picked from commit a0987672f0)	1 year ago
George Joseph	a81c7c7030	Add C++ Standard detection to configure and fix a new C++20 compile issue ... * The autoconf-archive package contains macros useful for detecting C++ standard and testing other C++ capabilities but that package was never included in the install_prereq script so many existing build environments won't have it. Even if it is installed, older versions won't newer C++ standards and will actually cause an error if you try to test for that version. To make it available for those environments, the ax_cxx_compile_stdcxx.m4 macro has copied from the latest release of autoconf-archive into the autoconf directory. * A convenience wrapper(ast_cxx_check_std) around ax_cxx_compile_stdcxx was also added so checking the standard version and setting the asterisk-specific PBX_ variables becomes a one-liner: `AST_CXX_CHECK_STD([std], [force_latest_std])`. Calling that with a version of `17` for instance, will set PBX_CXX17 to 0 or 1 depending on whether the current c++ compiler supports stdc++17. HAVE_CXX17 will also be 'defined" or not depending on the result. * C++ compilers hardly ever default to the latest standard they support. g++ version 14 for instance supports up to C++23 but only uses C++17 by default. If you want to use C++23, you have to add `-std=gnu++=23` to the g++ command line. If you set the second argument of AST_CXX_CHECK_STD to "yes", the macro will automatically keep the highest `-std=gnu++` value that worked and pass that to the Makefiles. * The autoconf-archive package was added to install_prereq for future use. * Updated configure.ac to use AST_CXX_CHECK_STD() to check for C++ versions 11, 14, 17, 20 and 23. * Updated configure.ac to accept the `--enable-latest-cxx-std` option which will set the second option to AST_CXX_CHECK_STD() to "yes". The default is "no". * ast_copy_string() in strings.h declares the 'sz' variable as volatile and does an `sz--` on it later. C++20 no longer allows the `++` and `--` increment and decrement operators to be used on variables declared as volatile however so that was changed to `sz -= 1`. (cherry picked from commit c193752ca4)	1 year ago
Naveen Albert	db2bcf3e19	chan_dahdi: Fix wrong channel state when RINGING recieved. ... Previously, when AST_CONTROL_RINGING was received by a DAHDI device, it would set its channel state to AST_STATE_RINGING. However, an analysis of the codebase and other channel drivers reveals RINGING corresponds to physical power ringing, whereas AST_STATE_RING should be used for audible ringback on the channel. This also ensures the correct device state is returned by the channel state to device state conversion. Since there seems to be confusion in various places regarding AST_STATE_RING vs. AST_STATE_RINGING, some documentation has been added or corrected to clarify the actual purposes of these two channel states, and the associated device state mapping. An edge case that prompted this fix, but isn't explicitly addressed here, is that of an incoming call to an FXO port. The channel state will be "Ring", which maps to a device state of "In Use", not "Ringing" as would be more intuitive. However, this is semantic, since technically, Asterisk is treating this the same as any other incoming call, and so "Ring" is the semantic state (put another way, Asterisk isn't ringing anything, like in the cases where channels are in the "Ringing" state). Since FXO ports don't currently support Call Waiting, a suitable workaround for the above would be to ignore the device state and instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state))) since it will be Ring if the FXO port is idle (but a call is ringing on it) and Up if the FXO port is actually in use. (In both cases, the device state would misleadingly be "In Use".) Resolves: #1029 (cherry picked from commit c5e7721341)	1 year ago
George Joseph	b3cc0c1026	Header fixes for compiling C++ source files ... A few tweaks needed to be done to some existing header files to allow them to be compiled when included from C++ source files. logger.h had declarations for ast_register_verbose() and ast_unregister_verbose() which caused C++ issues but those functions were actually removed from logger.c many years ago so the declarations were just removed from logger.h. (cherry picked from commit e8c5c3380a)	1 year ago
George Joseph	1324eb05c5	Add ability to pass arguments to unit tests from the CLI ... Unit tests can now be passed custom arguments from the command line. For example, the following command would run the "mytest" test in the "/main/mycat" category with the option "myoption=54" `CLI> test execute category /main/mycat name mytest options myoption=54` You can also pass options to an entire category... `CLI> test execute category /main/mycat options myoption=54` Basically, everything after the "options" keyword is passed verbatim to the test which must decide what to do with it. * A new API ast_test_get_cli_args() was created to give the tests access to the cli_args->argc and cli_args->argv elements. * Although not needed for the option processing, a new macro ast_test_validate_cleanup_custom() was added to test.h that allows you to specify a custom error message instead of just "Condition failed". * The test_skel.c was updated to demonstrate parsing options and the use of the ast_test_validate_cleanup_custom() macro. (cherry picked from commit b33acc8f86)	1 year ago
Kent	ef66694b7b	res_pjsip: Add new AOR option "qualify_2xx_only" ... Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify feature to mark a contact as available only if an OPTIONS request returns a 2XX response. If the option is not specified or is false, any response to the OPTIONS request marks the contact as available. UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only" option that can be set so that only 2XX responses to OPTIONS requests used to qualify a contact will mark the contact as available. (cherry picked from commit 0b30f546ba)	1 year ago
George Joseph	f5dbb87a48	res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" ... Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party. Resolves: #979 UserNote: The new "suppress_moh_on_sendonly" endpoint option can be used to prevent playing MOH back to a caller if the remote end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP. (cherry picked from commit 98510d4c75)	1 year ago
George Joseph	ec5387748c	res_pjsip: Move tenantid to end of ast_sip_endpoint ... The tenantid field was originally added to the ast_sip_endpoint structure at the end of the AST_DECLARE_STRING_FIELDS block. This caused everything after it in the structure to move down in memory and break ABI compatibility. It's now at the end of the structure as an AST_STRING_FIELD_EXTENDED. Given the number of string fields in the structure now, the initial string field allocation was also increased from 64 to 128 bytes. Resolves: #982 (cherry picked from commit 3dfac27ef2)	1 year ago
Thomas Guebels	4d7d5deab3	pjsip_transport_events: handle multiple addresses for a domain ... The key used for transport monitors was the remote host name for the transport and not the remote address resolved for this domain. This was problematic for domains returning multiple addresses as several transport monitors were created with the same key. Whenever a subsystem wanted to register a callback it would always end up attached to the first transport monitor with a matching key. The key used for transport monitors is now the remote address and port the transport actually connected to. Fixes: #932 (cherry picked from commit d1ed1018c4)	1 year ago