asterisk

Commit Graph

Author	SHA1	Message	Date
Artem Umerov	92d69897b7	logger.h: Fix build when AST_DEVMODE is not defined. Resolves: #1058	9 months ago
Sean Bright	032235bd29	dialplan_functions_doc.xml: Document PJSIP_MEDIA_OFFER's `media` argument. Resolves: #1023	9 months ago
Abdelkader Boudih	4b6d40bd66	samples: Use "asterisk" instead of "postgres" for username	9 months ago
Sean Bright	131682c2c5	manager: Add `<since>` tags for all AMI actions.	9 months ago
Steffen Arntz	d31b79b2f8	logger.c fix: malformed JSON template this typo was mentioned before, but never got fixed. https://community.asterisk.org/t/logger-cannot-log-long-json-lines-properly/87618/6	9 months ago
Sean Bright	e4635e817f	manager.c: Rename restrictedFile to is_restricted_file. Also correct the spelling of 'privileges.'	9 months ago
Abdelkader Boudih	7c40d82f02	res_config_pgsql: normalize database connection option with cel and cdr by supporting new options name	9 months ago
Stanislav Abramenkov	e95f97ad38	res_pjproject: Fix typo (OpenmSSL->OpenSSL) Fix typo (OpenmSSL->OpenSSL) mentioned by bkford in #972	9 months ago
George Joseph	7dc9d85f2b	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.	9 months ago
Allan Nathanson	1cb741df4a	config.c: retain leading whitespace before comments Configurations loaded with the ast_config_load2() API and later written out with ast_config_text_file_save2() will have any leading whitespace stripped away. The APIs should make reasonable efforts to maintain the content and formatting of the configuration files. This change retains any leading whitespace from comment lines that start with a ";". Resolves: https://github.com/asterisk/asterisk/issues/970	9 months ago
Ben Ford	e358ce0762	manager.c: Restrict ListCategories to the configuration directory. When using the ListCategories AMI action, it was possible to traverse upwards through the directories to files outside of the configured configuration directory. This action is now restricted to the configured directory and an error will now be returned if the specified file is outside of this limitation. Resolves: #GHSA-33x6-fj46-6rfh UserNote: The ListCategories AMI action now restricts files to the configured configuration directory.	10 months ago
Sean Bright	6b3dadfbe2	config.c: Fix off-nominal reference leak. This was identified and fixed by @Allan-N in #918 but it is an important fix in its own right. The fix here is slightly different than Allan's in that we just move the initialization of the problematic AO2 container to where it is first used. Fixes #1046	10 months ago
Abdelkader Boudih	cc2b5ec88b	normalize contrib/ast-db-manage/queue_log.ini.sample	10 months ago
George Joseph	79427c1ac1	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	afd15274e7	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
Stanislav Abramenkov	2e8d8c4a08	Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016 UserNote: Bundled pjproject has been upgraded to 2.15.1. For more information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.15.1	10 months ago
George Joseph	5beb61317a	.github: Set exit 0 in CherryPick and Recheck workflow Cleanup steps	10 months ago
George Joseph	6fa7c40b57	gcc14: Fix issues caught by gcc 14 * test_message.c: Fix segfaults caused by passing NULL as an sprintf fmt.	10 months ago
George Joseph	e677ec8473	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	2fa9dcd51c	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	21dba60ff2	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	865c8b8232	.github: Change the run name for OnPRStateChangedPriv	10 months ago
Jaco Kroon	9b45103bc8	res_odbc: release threads from potential starvation. Whenever a slot is freed up due to a failed connection, wake up a waiter before failing. In the case of a dead connection there could be waiters, for example, let's say two threads tries to acquire objects at the same time, with one in the cached connections, one will acquire the dead connection, and the other will enter into the wait state. The thread with the dead connection will clear up the dead connection, and then attempt a re-acquire (at this point there cannot be cached connections else the other thread would have received that and tried to clean up), as such, at this point we're guaranteed that either there are no waiting threads, or that the maxconnections - connection_cnt threads will attempt to re-acquire connections, and then either succeed, using those connections, or failing, and then signalling to release more waiters. Also fix the pointer log for ODBC handle %p dead which would always reflect NULL. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	10 months ago
Sperl Viktor	d832aae1d7	app_queue: indicate the paused state of a dynamically added member in queue_log. Fixes: #1021	11 months ago
George Joseph	9a1180c6af	Allow C++ source files (as extension .cc) in the main directory Although C++ files (as extension .cc) have been handled in the module directories for many years, the main directory was missing one line in its Makefile that prevented C++ files from being recognised there.	11 months ago
Alexey Khabulyak	ced3b41327	format_gsm.c: Added mime type Sometimes it's impossible to get a file extension from URL (eg. http://example.com/gsm/your) so we have to rely on content-type header. Currenly, asterisk does not support content-type for gsm format(unlike wav). Added audio/gsm according to https://www.rfc-editor.org/rfc/rfc4856.html	11 months ago
Maksim Nesterov	c50dea93a5	func_uuid: Add a new dialplan function to generate UUIDs This function is useful for uniquely identifying calls, recordings, and other entities in distributed environments, as well as for generating an argument for the AudioSocket application.	11 months ago
Sperl Viktor	7b7df5d30e	app_queue: allow dynamically adding a queue member in paused state. Fixes: #1007 UserNote: use the p option of AddQueueMember() for paused member state. Optionally, use the r(reason) option to specify a custom reason for the pause.	11 months ago
Naveen Albert	be2bea854e	chan_iax2: Add log message for rejected calls. Add a log message for a path that currently silently drops IAX2 frames without indicating that anything is wrong.	11 months ago
Maximilian Fridrich	45c5d39c58	chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams Currently, when a chan_pjsip channel receives a VIDUPDATE indication, an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is enabled on that endpoint. This restriction does not really make sense. Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as is the case with VP8, VP9, and H.265 streams). Resolves: #1013	11 months ago
Tinet-mucw	3ac5f82d49	audiohook.c: resolving the issue with audiohook both reading when packet loss on one side of the call When there is 0% packet loss on one side of the call and 15% packet loss on the other side, reading frame is often failed when reading direction_both audiohook. when read_factory available = 0, write_factory available = 320; i think write factory is usable read; because after reading one frame, there is still another frame that can be read together with the next read factory frame. Resolves: #851	11 months ago
Mike Pultz	424e4ee761	res_curl.conf.sample: clean up sample configuration and add new SSL options This update properly documents all the current configuration options supported by the curl implementation, including the new ssl_* options.	11 months ago
Viktor Litvinov	2599aa3be0	res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big Set Mark bit in rtp stream when timestamp skew is bigger than MAX_TIMESTAMP_SKEW. Fixes: #927	11 months ago
Alexey Vasilyev	e96d1cf2ae	res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF Fixes #1004	11 months ago
Mike Pultz	0c193b725b	manager.c: Add Processed Call Count to CoreStatus output This update adds the processed call count to the CoreStatus AMI Action responsie. This output is similar to the values returned by "core show channels" or "core show calls" in the CLI. UserNote: The current processed call count is now returned as CoreProcessedCalls from the CoreStatus AMI Action.	11 months ago
Mike Pultz	8e289af9de	func_curl.c: Add additional CURL options for SSL requests This patch adds additional CURL TLS options / options to support mTLS authenticated requests: * ssl_verifyhost - perform a host verification on the peer certificate (CURLOPT_SSL_VERIFYHOST) * ssl_cainfo - define a CA certificate file (CURLOPT_CAINFO) * ssl_capath - define a CA certificate directory (CURLOPT_CAPATH) * ssl_cert - define a client certificate for the request (CURLOPT_SSLCERT) * ssl_certtype - specify the client certificate type (CURLOPT_SSLCERTTYPE) * ssl_key - define a client private key for the request (CURLOPT_SSLKEY) * ssl_keytype - specify the client private key type (CURLOPT_SSLKEYTYPE) * ssl_keypasswd - set a password for the private key, if required (CURLOPT_KEYPASSWD) UserNote: The following new configuration options are now available in the res_curl.conf file, and the CURL() function: 'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo' (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert' (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE), 'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype', (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD). See the libcurl documentation for more details.	11 months ago
Naveen Albert	4f0a506c67	sig_analog: Fix regression with FGD and E911 signaling. Commit `466eb4a52b` introduced a regression which completely broke Feature Group D and E911 signaling, by removing the call to analog_my_getsigstr, which affected multiple switch cases. Restore the original behavior for all protocols except Feature Group C CAMA (MF), which is all that patch was attempting to target. Resolves: #993	11 months ago
James Terhune	d6b7554012	main/stasis_channels.c: Fix crash when setting a global variable with invalid UTF8 characters Add check for null value of chan before referencing it with ast_channel_name() Resolves: #999	11 months ago
George Joseph	e9f336b47b	res_stir_shaken: Allow sending Identity headers for unknown TNs Added a new option "unknown_tn_attest_level" to allow Identity headers to be sent when a callerid TN isn't explicitly configured in stir_shaken.conf. Since there's no TN object, a private_key_file and public_cert_url must be configured in the attestation or profile objects. Since "unknown_tn_attest_level" uses the same enum as attest_level, some of the sorcery macros had to be refactored to allow sharing the enum and to/from string conversion functions. Also fixed a memory leak in crypto_utils:pem_file_cb(). Resolves: #921 UserNote: You can now set the "unknown_tn_attest_level" option in the attestation and/or profile objects in stir_shaken.conf to enable sending Identity headers for callerid TNs not explicitly configured.	11 months ago
George Joseph	1f67825a91	res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T The suppress_moh_on_sendonly endpoint option should have been defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES in the alembic script instead of OPT_YESNO_T and YESNO_VALUES. Also updated contrib/ast-db-manage/README.md to indicate that AST_BOOL_VALUES should always be used and provided an example. Resolves: #995	11 months ago
George Joseph	03ab004e26	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.	11 months ago
Sean Bright	1c0fcd583d	res_pjsip.c: Fix Contact header rendering for IPv6 addresses. Fix suggested by @nvsystems. Fixes #985	11 months ago
chrsmj	c037fc7fae	samples: remove and/or change some wiki mentions Cleaned some dead links. Replaced word wiki with either docs or link to https://docs.asterisk.org/ Resolves: #974	11 months ago
George Joseph	aa736dac12	func_pjsip_aor/contact: Fix documentation for contact ID Clarified the use of the contact ID returned from PJSIP_AOR. Resolves: #990	11 months ago
George Joseph	144033db31	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	11 months ago
Thomas Guebels	184a30efac	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	11 months ago
Naveen Albert	34662cf0ea	func_evalexten: Add EVAL_SUB function. This adds an EVAL_SUB function, which is similar to the existing EVAL_EXTEN function but significantly more powerful, as it allows executing arbitrary dialplan and capturing its return value as the function's output. While EVAL_EXTEN should be preferred if it is possible to use it, EVAL_SUB can be used in a wider variety of cases and allows arbitrary computation to be performed in a dialplan function call, leveraging the dialplan. Resolves: #951	11 months ago
George Joseph	2717068e1b	res_srtp: Change Unsupported crypto suite msg from verbose to debug There's really no point in spamming logs with a verbose message for every unsupported crypto suite an older client may send in an SDP. If none are supported, there will be an error or warning.	12 months ago
Ben Ford	9f1f3703c9	Add res_pjsip_config_sangoma external module. Adds res_pjsip_config_sangoma as an external module that can be downloaded via menuselect. It lives under the Resource Modules section.	12 months ago
Ben Ford	4616408b09	app_mixmonitor: Add 'D' option for dual-channel audio. Adds the 'D' option to app_mixmonitor that interleaves the input and output frames of the channel being recorded in the monitor output frame. This allows for two streams in the recording: the transmitted audio and the received audio. The 't' and 'r' options are compatible with this. Fixes: #945 UserNote: The MixMonitor application now has a new 'D' option which interleaves the recorded audio in the output frames. This allows for stereo recording output with one channel being the transmitted audio and the other being the received audio. The 't' and 't' options are compatible with this.	12 months ago

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