4426 Commits (8149554e9004c2a37e2fa8ca26dbe19d99241229)

Author	SHA1	Message	Date
Tinet-mucw	8149554e90	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	4 months ago
George Joseph	c873f2ae7e	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	5a3164c0b2	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	3a5ffe2842	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	8f1982c4d6	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	6 months ago
Naveen Albert	2ced79259a	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.	6 months ago
George Joseph	6fa0e264df	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.	6 months ago
Albrecht Oster	a9cd7f9b8d	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	6 months ago
George Joseph	f302c116b4	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().	6 months ago
Sean Bright	9fdf8597cb	channel: Deprecate `ast_moh_cleanup(...)`. ... We don't want anyone calling it but the channel destructor.	7 months ago
George Joseph	3d5ae0b5e1	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/	7 months ago
mkmer	edadca7151	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	7 months ago
Florent CHAUVEAU	a5bc39fa32	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,*,#...).	7 months ago
Norm Harrison	6ef4a2d008	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' ... Co-authored-by: Florent CHAUVEAU <florentch@pm.me>	7 months ago
Norm Harrison	4e546c35c7	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>	7 months ago
George Joseph	076799be15	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	8 months ago
George Joseph	d558818ec1	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	9 months ago
Holger Hans Peter Freyther	9fc631d028	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.	9 months ago
Sean Bright	e200f51ea4	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.	9 months ago
George Joseph	32e6a30952	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().	9 months ago
Allan Nathanson	217e696288	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	10 months ago
George Joseph	3868f7cb47	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.	10 months ago
Artem Umerov	ae106fadfe	logger.h: Fix build when AST_DEVMODE is not defined. ... Resolves: #1058	10 months ago
George Joseph	71a2e8c599	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.	10 months ago
George Joseph	62ec48cf1c	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`.	10 months ago
Naveen Albert	5ed7f4d152	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	10 months ago
George Joseph	a33505b334	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.	10 months ago
George Joseph	92e5ed5a3a	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.	10 months ago
Kent	3dee037446	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.	10 months ago
George Joseph	02e2000653	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.	12 months ago
George Joseph	3e9b00a115	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	12 months ago
Thomas Guebels	5d26fef407	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	12 months ago
Allan Nathanson	97770a947c	dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin ... The dnsmgr_refresh() function checks to see if the IP address associated with a name/service has changed. The gotcha is that the ast_get_ip_or_srv() function only returns the first IP address returned by the DNS query. If there are multiple IPs associated with the name and the returned order is not consistent (e.g. with DNS round-robin) then the other IP addresses are not included in the comparison and the entry is flagged as changed even though the IP is still valid. Updated the code to check all IP addresses and flag a change only if the original IP is no longer valid. Resolves: #924	1 year ago
Sean Bright	e76f671810	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. ... Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922	1 year ago
Mike Bradeen	055031dfcb	res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch ... When Asterisk sends an offer to Bob that includes 48K and 8K codecs with matching 4733 offers, Bob may want to use the 48K audio codec but can not accept 48K digits and so negotiates for a mixed set. Asterisk will now check Bob's offer to make sure Bob has indicated this is acceptible and if not, will use Bob's preference. Fixes: #847	1 year ago
Mike Bradeen	7a7926a0d9	Prepare for Asterisk 23	1 year ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event ... A stasis event is now produced when using the TONE_DETECT dialplan function. This event is published over ARI using the ChannelToneDetected event. This change does not make it available over AMI. Fixes: #811 UserNote: Setting the TONE_DETECT dialplan function on a channel in ARI will now cause a ChannelToneDetected ARI event to be raised when the specified tone is detected.	1 year ago
Mike Bradeen	52bdfe09ab	res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits ... After change made in 624f509 to add support for non 8K RFC 4733/2833 digits, Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of the negotiated codec(s). This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec. A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to be re-written to allow for these scenarios. Fixes: #776	1 year ago
Sean Bright	8b5ddfee5e	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. ... Fixes #785	1 year ago
Sean Bright	f3449f3370	file.h: Rename function argument to avoid C++ keyword clash. ... Fixes #744	1 year ago
George Joseph	85241bd229	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
Mike Bradeen	3b4d2a6506	rtp_engine: add support for multirate RFC2833 digits ... Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs. Asterisk currently treats RFC2833 Digits as a single rtp payload type with a fixed bitrate of 8K. This change would expand that to 8, 16, 24 and 32K. This requires checking the offered rtp types for any of these bitrates and then adding an offer for each (if configured for RFC2833.) DTMF generation must also be changed in order to look at the current outbound codec in order to generate appropriately timed rtp. For cases where no outgoing audio has yet been sent prior to digit generation, Asterisk now has a concept of a 'preferred' codec based on offer order. On inbound calls Asterisk will mimic the payload types of the RFC2833 digits. On outbound calls Asterisk will choose the next free payload types starting with 101. UserNote: No change in configuration is required in order to enable this feature. Endpoints configured to use RFC2833 will automatically have this enabled. If the endpoint does not support this, it should not include it in the SDP offer/response. Resolves: #699	1 year ago
Naveen Albert	1f1c5bb902	logger: Add unique verbose prefixes for levels 5-10. ... Add unique verbose prefixes for levels higher than 4, so that these can be visually differentiated from each other. Resolves: #721	2 years ago
Naveen Albert	01ff5c8140	loader.c: Allow dependent modules to be unloaded recursively. ... Because of the (often recursive) nature of module dependencies in Asterisk, hot swapping a module on the fly is cumbersome if a module is depended on by other modules. Currently, dependencies must be popped manually by unloading dependents, unloading the module of interest, and then loading modules again in reverse order. To make this easier, the ability to do this recursively in certain circumstances has been added, as an optional extension to the "module refresh" command. If requested, Asterisk will check if a module that has a positive usecount could be unloaded safely if anything recursively dependent on it were unloaded. If so, it will go ahead and unload all these modules and load them back again. This makes hot swapping modules that provide dependencies much easier. Resolves: #474 UserNote: In certain circumstances, modules with dependency relations can have their dependents automatically recursively unloaded and loaded again using the "module refresh" CLI command or the ModuleLoad AMI command.	2 years ago
George Joseph	8e119a72f0	tcptls/iostream: Add support for setting SNI on client TLS connections ... If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello.	2 years ago
Naveen Albert	c1b69d460d	callerid.c: Parse previously ignored Caller ID parameters. ... Commit f2f397c1a8 previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681	2 years ago
George Joseph	57ce1fe01b	logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT ... If you're tracing a large function that may call another function multiple times in different circumstances, it can be difficult to see from the trace output exactly which location that function was called from. There's no good way to automatically determine the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT simply print out a trace line before and after the call. The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is that SCOPE_CALL ignores the function's return value (if any) where SCOPE_CALL_WITH_RESULT allows you to specify the type of the function's return value so it can be assigned to a variable. SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT and the "int" return type.	2 years ago
Sperl Viktor	0ab4a5ef6b	res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address. ... Add a new identify_by option to res_pjsip_endpoint_identifier_ip called 'transport' this matches endpoints based on the bound ip address (local) instead of the 'ip' option, which matches on the source ip address (remote). UserNote: set identify_by=transport for the pjsip endpoint. Then use the existing 'match' option and the new 'transport' option of the identify. Fixes: #672	2 years ago
Sperl Viktor	895ab9d798	res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI ... Add ability to match against PJSIP request URI. UserNote: this new feature let users match endpoints based on the indound SIP requests' URI. To do so, add 'request_uri' to the endpoint's 'identify_by' option. The 'match_request_uri' option of the identify can be an exact match for the entire request uri, or a regular expression (between slashes). It's quite similar to the header identifer. Fixes: #599	2 years ago