4322 Commits (18)

Author	SHA1	Message	Date
Igor Goncharovsky	de745157ca	res_pjsip_rfc3326: Add SIP causes support for RFC3326 ... Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	2 years ago
George Joseph	345ff2d8ee	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	8067229418	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit 62745013a4. Reason for revert: Issue when transcoding to/from g722 Change-Id: I1665a5442bfb6d7bfa06fdcea3374f4581395b4a	2 years ago
Naveen Albert	811584ded3	json.h: Add ast_json_object_real_get. ... json.h contains macros to get a string and an integer from a JSON object. However, the macro to do this for JSON reals is missing. This adds that. ASTERISK-30361 #close Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a	2 years ago
George Joseph	62745013a4	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	40ed776c40	res_pjsip_session: Use Caller ID for extension matching. ... Currently, there is no Caller ID available to us when checking for an extension match when handling INVITEs. As a result, extension patterns that depend on the Caller ID are not matched and calls may be incorrectly rejected. The Caller ID is not available because the supplement that adds Caller ID to the session does not execute until after this check. Supplement callbacks cannot yet be executed at this point since the session is not yet in the appropriate state. To fix this without impacting existing behavior, the Caller ID number is now retrieved before attempting to pattern match. This ensures pattern matching works correctly and there is no behavior change to the way supplements are called. ASTERISK-28767 #close Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7	2 years ago
Peter Fern	b1f1556922	streams: Ensure that stream is closed in ast_stream_and_wait on error ... When ast_stream_and_wait returns an error (for example, when attempting to stream to a channel after hangup) the stream is not closed, and callers typically do not check the return code. This results in leaking file descriptors, leading to resource exhaustion. This change ensures that the stream is closed in case of error. ASTERISK-30198 #close Reported-by: Julien Alie Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803	2 years ago
Michael Kuron	d73f22268f	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip ... chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333	2 years ago
Naveen Albert	0fb48fb751	rtp_engine.h: Update examples using ast_format_set. ... This file includes some doxygen comments referencing ast_format_set. This is an obsolete API that was removed years back, but documentation was not fully updated to reflect that. These examples are updated to the current way of doing things (using the format cache). ASTERISK-30327 #close Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7	2 years ago
George Joseph	5741569c5e	pjsip_transport_events: Fix possible use after free on transport ... It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	2 years ago
Mike Bradeen	76d10ea668	manager: prevent file access outside of config dir ... Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404	2 years ago
Henning Westerholt	093593a7c8	res_pjsip: return all codecs on a re-INVITE without SDP ... Currently chan_pjsip on receiving a re-INVITE without SDP will only return the codecs that are previously negotiated and not offering all enabled codecs. This causes interoperability issues with different equipment (e.g. from Cisco) for some of our customers and probably also in other scenarios involving 3PCC infrastructure. According to RFC 3261, section 14.2 we SHOULD return all codecs on a re-INVITE without SDP The PR proposes a new parameter to configure this behaviour: all_codecs_on_empty_reinvite. It includes the code, documentation, alembic migrations, CHANGES file and example configuration additions. ASTERISK-30193 #close Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148	2 years ago
Mike Bradeen	4fd2f07ff9	audiohook: add directional awareness ... Add enum to allow setting optional direction. If set to only one direction, only feed matching-direction frames to the associated slin factory. This prevents mangling the transcoder on non-mixed frames when the READ and WRITE frames would have otherwise required it. Also removes the need to mute or discard the un-wanted frames as they are no longer added in the first place. res_stasis_snoop is changed to use this addition to set direction on audiohook based on spy direction. If no direction is set, the ast_audiohook_init will init this enum to BOTH which maintains existing functionality. ASTERISK-30252 Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb	2 years ago
Naveen Albert	faca1fcfd3	cdr: Allow bridging and dial state changes to be ignored. ... Allows bridging, parking, and dial messages to be globally ignored for all CDRs such that only a single CDR record is generated per channel. This is useful when CDRs should endure for the lifetime of an entire channel and bridging and dial updates in the dialplan should not result in multiple CDR records being created for the call. With the ignore bridging option, bridging changes have no impact on the channel's CDRs. With the ignore dial state option, multiple Dials and their outcomes have no impact on the channel's CDRs. The last disposition on the channel is preserved in the CDR, so the actual disposition of the call remains available. These two options can reduce the amount of "CDR hacks" that have hitherto been necessary to ensure that CDR was not "spoiled" by these messages if that was undesired, such as putting a dummy optimization-disabled local channel between the caller and the actual call and putting the CDR on the channel in the middle to ensure that CDR would persist for the entire call and properly record start, answer, and end times. Enabling these options is desirable when calls correspond to the entire lifetime of channels and the CDR should reflect that. Current default behavior remains unchanged. ASTERISK-30091 #close Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832	2 years ago
Maximilian Fridrich	675bed5f2c	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	2 years ago
George Joseph	96bf50b146	manager.h: Bump version to 7.0.3 ... Change-Id: I5c365e141769f8ce2a1aeb4ae2cdb9842dd2fd5b	2 years ago
Philip Prindeville	ddc8f6967f	test: initialize capture structure before freeing ... ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	2 years ago
Maximilian Fridrich	479ce16eba	res_pjsip: Add 100rel option "peer_supported". ... This patch adds a new option to the 100rel parameter for pjsip endpoints called "peer_supported". When an endpoint with this option receives an incoming request and the request indicated support for the 100rel extension, then Asterisk will send 1xx responses reliably. If the request did not indicate 100rel support, Asterisk sends 1xx responses normally. ASTERISK-30158 Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad	2 years ago
George Joseph	4ab3cd6f86	res_geolocation: Fix segfault when there's an empty element ... Fixed a segfault caused by var_list_from_loc_info() encountering an empty location info element. Fixed an issue in ast_strsep() where a value with only whitespace wasn't being preserved. Fixed an issue in ast_variable_list_from_quoted_string() where an empty value was considered a failure. ASTERISK-30215 Reported by: Dan Cropp Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a	2 years ago
Ben Ford	1f685d6969	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	2 years ago
Philip Prindeville	bb8b2259bc	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	2 years ago
Philip Prindeville	0972106d51	test: Add coverage for res_crypto ... We're validating the following functionality: encrypting a block of data with RSA decrypting a block of data with RSA signing a block of data with RSA verifying a signature with RSA encrypting a block of data with AES-ECB encrypting a block of data with AES-ECB as well as accessing test keys from the keystore. ASTERISK-30045 #close Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d	2 years ago
Philip Prindeville	53f8cc1d49	res_crypto: make keys reloadable on demand for testing ... ASTERISK-30045 Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8	2 years ago
Philip Prindeville	aeb23fe2a7	main/utils: allow checking for command in $PATH ... ASTERISK-30037 Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf	2 years ago
Philip Prindeville	f2cd48e9a8	test: Add ability to capture child process output ... ASTERISK-30037 Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2	2 years ago
George Joseph	0e9076df7b	res_geolocation: Add two new options to GEOLOC_PROFILE ... Added an 'a' option to the GEOLOC_PROFILE function to allow variable lists like location_info_refinement to be appended to instead of replacing the entire list. Added an 'r' option to the GEOLOC_PROFILE function to resolve all variables before a read operation and after a Set operation. Added a few missing parameters to the ones allowed for writing with GEOLOC_PROFILE. Fixed a bug where calling GEOLOC_PROFILE to read a parameter might actually update the profile object. Cleaned up XML documentation a bit. ASTERISK-30190 Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c	2 years ago
George Joseph	261051ecec	res_geolocation: Allow location parameters on the profile object ... You can now specify the location object's format, location_info, method, location_source and confidence parameters directly on a profile object for simple scenarios where the location information isn't common with any other profiles. This is mutually exclusive with setting location_reference on the profile. Updated appdocsxml.dtd to allow xi:include in a configObject element. This makes it easier to link to complete configOptions in another object. This is used to add the above fields to the profile object without having to maintain the option descriptions in two places. ASTERISK-30185 Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569	2 years ago
George Joseph	372f5c56e1	res_geolocation: Add profile parameter suppress_empty_ca_elements ... Added profile parameter "suppress_empty_ca_elements" that will cause Civic Address elements that are empty to be suppressed from the outgoing PIDF-LO document. Fixed a possible SEGV if a sub-parameter value didn't have a value. ASTERISK-30177 Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd	2 years ago
Naveen Albert	8041bd12c1	cli: Prevent assertions on startup from bad ao2 refs. ... If "core show channels" is run before startup has completed, it is possible for bad ao2 refs to occur because the system is not yet fully initialized. This will lead to an assertion failing. To prevent this, initialization of CLI builtins is moved to be later along in the main load sequence. Core CLI commands are loaded at the same time, but channel-related commands are loaded later on. ASTERISK-29846 #close Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6	2 years ago
Joshua C. Colp	f97e9aed85	pjsip: Add TLS transport reload support for certificate and key. ... This change adds support using the pjsip_tls_transport_restart function for reloading the TLS certificate and key, if the filenames remain unchanged. This is useful for Let's Encrypt and other situations. Note that no restart of the transport will occur if the certificate and key remain unchanged. ASTERISK-30186 Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0	2 years ago
Sean Bright	57972bac38	channel.h: Remove redundant declaration. ... The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set() for us, so no need to declare it separately. Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a	2 years ago
Naveen Albert	6000f9963c	features: Add transfer initiation options. ... Adds additional control options over the transfer feature functionality to give users more control in how the transfer feature sounds and works. First, the "transfer" sound that plays when a transfer is initiated can now be customized by the user in features.conf, just as with the other transfer sounds. Secondly, the user can now specify the transfer extension in advance by using the TRANSFER_EXTEN variable. If a valid extension is contained in this variable, the call will automatically be transferred to this destination. Otherwise, it will fall back to collecting the extension from the user as is always done now. ASTERISK-29899 #close Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3	2 years ago
Naveen Albert	9126f9bd2b	general: Very minor coding guideline fixes. ... Fixes a few coding guideline violations: * Use of C99 comments * Opening brace on same line as function prototype ASTERISK-30163 #close Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7	2 years ago
George Joseph	dfedb13eca	res_geolocation: Address user issues, remove complexity, plug leaks ... * Added processing for the 'confidence' element. * Added documentation to some APIs. * removed a lot of complex code related to the very-off-nominal case of needing to process multiple location info sources. * Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes one eprofile instead of a datastore of multiples. * Plugged a huge leak in XML processing that arose from insufficient documentation by the libxml/libxslt authors. * Refactored stylesheets to be more efficient. * Renamed 'profile_action' to 'profile_precedence' to better reflect it's purpose. * Added the config option for 'allow_routing_use' which sets the value of the 'Geolocation-Routing' header. * Removed the GeolocProfileCreate and GeolocProfileDelete dialplan apps. * Changed the GEOLOC_PROFILE dialplan function as follows: * Removed the 'profile' argument. * Automatically create a profile if it doesn't exist. * Delete a profile if 'inheritable' is set to no. * Fixed various bugs and leaks * Updated Asterisk WiKi documentation. ASTERISK-30167 Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0	2 years ago
Naveen Albert	8fddef679c	chan_dahdi: Fix buggy and missing Caller ID parameters ... There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15	2 years ago
George Joseph	cdbd973ecd	Geolocation: chan_pjsip Capability Preview ... This commit adds res_pjsip_geolocation which gives chan_pjsip the ability to use the core geolocation capabilities. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30128 Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9	2 years ago
George Joseph	f491355384	Geolocation: Core Capability Preview ... This commit adds res_geolocation which creates the core capabilities to manipulate Geolocation information on SIP INVITEs. An upcoming commit will add res_pjsip_geolocation which will allow the capabilities to be used with the pjsip channel driver. This commit message is intentionally short because this isn't a simple capability. See the documentation at https://wiki.asterisk.org/wiki/display/AST/Geolocation for more information. THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON USER FEEDBACK! ASTERISK-30127 Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303	2 years ago
Naveen Albert	0fe1a581e5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	2 years ago
George Joseph	1ed15e1815	Geolocation: Base Asterisk Prereqs ... * Added ast_variable_list_from_quoted_string() Parse a quoted string into an ast_variable list. * Added ast_str_substitute_variables_full2() Perform variable/function/expression substitution on an ast_str. * Added ast_strsep_quoted() Like ast_strsep except you can specify a specific quote character. Also added unit test. * Added ast_xml_find_child_element() Find a direct child element by name. * Added ast_xml_doc_dump_memory() Dump the specified document to a buffer * ast_datastore_free() now checks for a NULL datastore before attempting to destroy it. Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec	2 years ago
Naveen Albert	c9955d5fcd	db: Notify user if deleted DB entry didn't exist. ... Currently, if using the CLI to delete a DB entry, "Database entry removed" is always returned, regardless of whether or not the entry actually existed in the first place. This meant that users were never told if entries did not exist. The same issue occurs if trying to delete a DB key using AMI. To address this, new API is added that is more stringent in deleting values from AstDB, which will not return success if the value did not exist in the first place, and will print out specific error details if available. ASTERISK-30001 #close Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18	2 years ago
Kevin Harwell	0ddbf6bc45	res_pjsip: allow TLS verification of wildcard cert-bearing servers ... Rightly the use of wildcards in certificates is disallowed in accordance with RFC5922. However, RFC2818 does make some allowances with regards to their use when using subject alt names with DNS name types. As such this patch creates a new setting for TLS transports called 'allow_wildcard_certs', which when it and 'verify_server' are both enabled allows DNS name types, as well as the common name that start with '*.' to match as a wildcard. For instance: *.example.com will match for: foo.example.com Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc... And the starting wildcard only matches for a single level. For instance: *.example.com will NOT match for: foo.bar.example.com The new setting is disabled by default. ASTERISK-30072 #close Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4	2 years ago
Naveen Albert	6720caa29c	pbx: Add helper function to execute applications. ... Finding an application and executing it if found is a common task throughout Asterisk. This adds a helper function around pbx_exec to do this, to eliminate redundant code and make it easier for modules to substitute variables and execute applications by name. ASTERISK-30061 #close Change-Id: Ifee4d2825df7545fb515d763d393065675140c84	2 years ago
Naveen Albert	193688dd23	loader: Prevent deadlock using tab completion. ... If tab completion using ast_module_helper is attempted during startup, deadlock will ensue because the CLI will attempt to lock the module list while it is already locked by the loader. This causes deadlock because when the loader tries to acquire the CLI lock, they are blocked on each other. Waiting for startup to complete is not feasible because the CLI lock is acquired while waiting, so deadlock will ensure regardless of whether or not a lock on the module list is attempted. To prevent deadlock, we immediately abort if tab completion is attempted on the module list before Asterisk is fully booted. ASTERISK-30039 #close Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d	2 years ago
Moritz Fain	aaa14d3c7d	ari: expose channel driver's unique id to ARI channel resource ... This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	2 years ago
George Joseph	ad6af63895	GCC12: Fixes for 16+ ... Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570	2 years ago
Naveen Albert	79689d9df8	func_evalexten: Extension evaluation function. ... This adds the EVAL_EXTEN function, which may be used to retrieve the variable-substituted data at any extension. ASTERISK-29486 Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432	2 years ago
Mark Petersen	16e59db514	chan_pjsip: add allow_sending_180_after_183 option ... added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	2 years ago
Kevin Harwell	2fb8667908	res_aeap & res_speech_aeap: Add Asterisk External Application Protocol ... Add framework to connect to, and read and write protocol based messages from and to an external application using an Asterisk External Application Protocol (AEAP). This has been divided into several abstractions: 1. transport - base communication layer (currently websocket only) 2. message - AEAP description and data (currently JSON only) 3. transaction - links/binds requests and responses 4. aeap - transport, message, and transaction handler/manager This patch also adds an AEAP implementation for speech to text. Existing speech API callbacks for speech to text have been completed making it possible for Asterisk to connect to a configured external translator service and provide audio for STT. Results can also be received from the external translator, and made available as speech results in Asterisk. Unit tests have also been created that test the AEAP framework, and also the speech to text implementation. ASTERISK-29726 #close Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2	2 years ago
Sean Bright	2587e58e05	config.h: Don't use C++ keywords as argument names. ... ASTERISK-30021 #close Change-Id: I70eb59b782a4946b979942e21422746b7563029c	2 years ago
Ben Ford	11accf8064	AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. ... Adds a new configuration option, stir_shaken_profile, in pjsip.conf that can be specified on a per endpoint basis. This option will reference a stir_shaken_profile that can be configured in stir_shaken.conf. The type of this option must be 'profile'. The stir_shaken option can be specified on this object with the same values as before (attest, verify, on), but it cannot be off since having the profile itself implies wanting STIR/SHAKEN support. You can also specify an ACL from acl.conf (along with permit and deny lines in the object itself) that will be used to limit what interfaces Asterisk will attempt to retrieve information from when reading the Identity header. ASTERISK-29476 Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406	2 years ago