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Note that the Asterisk 19.* series is EOL and this package will be scheduled for deletion in one to two quarters. pkgsrc changes: - MKPIE_SUPPORTED=NO -- eol, so not worth effort to fix - various new/obsoleted config files / docs - new/obsoleted features + app_sf + func_evalexten + func_export + func_json + res_ari_mailboxes + res_geolocation + res_mwi_external + res_mwi_external_ami + res_pjsip_geolocation + res_pjsip_rfc3329 + res_speech_aeap + res_stasis_playback Change Log for Release 19.8.1 ======================================== Summary: ---------------------------------------- - apply_patches: Use globbing instead of file/sort. - bundled_pjproject: Backport 2 SSL patches from upstream - bundled_pjproject: Backport security fixes from pjproject 2.13.1 - apply_patches: Sort patch list before applying Closed Issues: ---------------------------------------- - #188: [improvement]: pjsip: Upgrade bundled version to pjproject 2.13.1 #187 - #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying - #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport Commits By Author: ---------------------------------------- - ### George Joseph (3): - apply_patches: Sort patch list before applying - bundled_pjproject: Backport security fixes from pjproject 2.13.1 - bundled_pjproject: Backport 2 SSL patches from upstream - ### Sean Bright (1): - apply_patches: Use globbing instead of file/sort. ----- ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.7.0 to Asterisk 19.8.0 ------------ ------------------------------------------------------------------------------ cdr ------------------ * Two new options have been added which allow bridging and dial state changes to be ignored in CDRs, which can be useful if a single CDR is desired for a channel. res_pjsip ------------------ * Added options "security_negotiation" and "security_mechanisms" to pjsip endpoints and registrations. "security_negotiation" can be set to "no" (default) or "mediasec", and "security_mechanisms" can be a list of comma-separated security_mechanisms in the form defined by RFC 3329 section 2.2. * A new option named "all_codecs_on_empty_reinvite" has been added to the global section. When this option is enabled, on reception of a re-INVITE without SDP, Asterisk will send an SDP offer in the 200 OK response containing all configured codecs on the endpoint, instead of simply those that have already been negotiated. RFC 3261 specifies this as a SHOULD requirement. The default value is "off". res_pjsip_logger ------------------ * SIP messages can now be filtered by SIP request method (INVITE, CANCEL, ACK, BYE, REGISTER, OPTION, SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE), allowing for more granular debugging to be done in the CLI. This applies to requests but not responses. res_pjsip_notify ------------------ * Allows using the config options in pjsip_notify.conf from AMI actions as with the existing CLI commands. res_tonedetect ------------------ * The TONE_DETECT function now supports detection of audible ringback tone using the p option. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.6.0 to Asterisk 19.7.0 ------------ ------------------------------------------------------------------------------ New EXPORT function ------------------ * A new function, EXPORT, allows writing variables and functions on other channels, the complement of the IMPORT function. app_amd ------------------ * An audio file to play during AMD processing can now be specified to the AMD application or configured in the amd.conf configuration file. app_bridgewait ------------------ * Adds the n option to not answer the channel when the BridgeWait application is called. features ------------------ * The Bridge application now has the n "no answer" option that can be used to prevent the channel from being automatically answered prior to bridging. func_strings ------------------ * Three new functions, TRIM, LTRIM, and RTRIM, are now available for trimming leading and trailing whitespace. res_pjsip ------------------ * A new option named "peer_supported" has been added to the endpoint option 100rel. When set to this option, Asterisk sends provisional responses reliably if the peer supports it. If the peer does not support reliable provisional responses, Asterisk sends them normally. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.6.0 to Asterisk 19.7.0 ------------ ------------------------------------------------------------------------------ Transfer feature ------------------ * The following capabilities have been added to the transfer feature: - The transfer initiation announcement prompt can now be customized in features.conf. - The TRANSFER_EXTEN variable now can be set on the transferer's channel in order to allow the transfer function to automatically attempt to go to the extension contained in this variable, if it exists. The transfer context behavior is not changed (TRANSFER_CONTEXT is used if it exists; otherwise the default context is used). app_confbridge ------------------ * Adds the end_marked_any option which can be used to kick users from a conference after any marked user leaves (including marked users). locks ------------------ * A new AMI event, DeadlockStart, is now available when Asterisk is compiled with DETECT_DEADLOCKS, and can indicate that a deadlock has occured. res_geolocation ------------------ * Added 4 built-in profiles: "<prefer_config>" "<discard_config>" "<prefer_incoming>" "<discard_incoming>" The profiles are empty except for having their precedence set. 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. 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. 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. res_musiconhold_answeredonly ------------------ * This change adds an option, answeredonly, that will prevent music on hold on channels that are not answered. res_pjsip ------------------ * TLS transports in res_pjsip can now reload their TLS certificate and private key files, provided the filename of them has not changed. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.5.0 to Asterisk 19.6.0 ------------ ------------------------------------------------------------------------------ res_geolocation ------------------ * * 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. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.5.0 to Asterisk 19.6.0 ------------ ------------------------------------------------------------------------------ chan_dahdi ------------------ * A POLARITY function is now available that allows getting or setting the polarity on a channel from the dialplan. db ------------------ * The DBPrefixGet AMI action now allows retrieving all of the DB keys beginning with a particular prefix. res_cliexec ------------------ * A new CLI command, dialplan exec application, has been added which allows dialplan applications to be executed at the CLI, useful for some quick testing without needing to write dialplan. res_geolocation ------------------ * Added res_geolocation which creates the core capabilities to manipulate Geolocation information on SIP INVITEs. res_pjsip ------------------ * A new transport option 'allow_wildcard_certs' has been added that when it and 'verify_server' are both set to 'yes', enables verification against wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS for TLS transport types. Names must start with the wildcard. Partial wildcards, e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only match against a single level meaning '*.example.com' matches 'foo.example.com', but not 'foo.bar.example.com'. res_pjsip_geolocation ------------------ * Added res_pjsip_geolocation which gives chan_pjsip the ability to use the core geolocation capabilities. res_pjsip_header_funcs ------------------ * Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request. Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------ ------------------------------------------------------------------------------ app_confbridge ------------------ * Added the hear_own_join_sound option to the confbridge user profile to control who hears the sound_join audio file. When set to 'yes' the user entering the conference and the participants already in the conference will hear the sound_join audio file. When set to 'no' the user entering the conference will not hear the sound_join audio file, but the participants already in the conference will hear the sound_join audio file. * Adds the CONFBRIDGE_CHANNELS function which can be used to retrieve a list of channels in a ConfBridge, optionally filtered by a particular category. This list can then be used with functions like SHIFT, POP, UNSHIFT, etc. app_queue ------------------ * The m option now allows an override music on hold class to be specified for the Queue application within the dialplan. app_voicemail ------------------ * The r option has been added, which prevents deletion of messages from VoiceMailMain, which can be useful for shared mailboxes. ari ------------------ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) to ARI channel resources as 'protocol_id'. ASTERISK-30027 chan_dahdi ------------------ * Previously, cadences were appended on dahdi restart, rather than reloaded. This prevented cadences from being updated and maxed out the available cadences if reloaded multiple times. This behavior is fixed so that reloading cadences is idempotent and cadences can actually be reloaded. chan_pjsip ------------------ * added global config option "allow_sending_180_after_183" Allow Asterisk to send 180 Ringing to an endpoint after 183 Session Progress has been send. If disabled Asterisk will instead send only a 183 Session Progress to the endpoint. * Hook flash events can now be sent on a PJSIP channel if requested to do so. chan_sip ------------------ * Session timers get removed on UPDATE Fix if Asterisk receives a SIP REFER with Session-Timers UAC that Asterisk maintains Session-Timers when sending UPDATE request cli ------------------ * A new CLI command 'dialplan eval function' has been added which allows users to test the behavior of dialplan function calls directly from the CLI. func_db ------------------ * The function DB_KEYCOUNT has been added, which returns the cardinality of the keys at a specified prefix in AstDB, i.e. the number of keys at a given prefix. func_evalexten ------------------ * This adds the EVAL_EXTEN function which may be used to evaluate data at dialplan extensions. res_agi ------------------ * Agi command 'exec' can now be enabled to evaluate dialplan functions and variables by setting the variable AGIEXECFULL to yes. res_parking ------------------ * An m option to Park and ParkAndAnnounce now allows specifying a music on hold class override. stasis_channels ------------------ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) to ARI channel resources as 'protocol_id'. ASTERISK-30027 ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.3.1 to Asterisk 19.3.2 ------------ ------------------------------------------------------------------------------ func_odbc ------------------ * A SQL_ESC_BACKSLASHES dialplan function has been added which escapes backslashes. Usage of this is dependent on whether the database in use can use backslashes to escape ticks or not. If it can, then usage of this prevents a broken SQL query depending on how the SQL query is constructed. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.2.0 to Asterisk 19.3.0 ------------ ------------------------------------------------------------------------------ ami ------------------ * AMI events can now be globally disabled using the disabledevents [general] setting. app_mf ------------------ * Adds an option to ReceiveMF to cap the number of digits read at a user-specified maximum. app_queue ------------------ * Load queues and members from Realtime for AMI actions: QueuePause, QueueStatus and QueueSummary, Applications: PauseQueueMember and UnpauseQueueMember. * Added a new AMI action: QueueWithdrawCaller This AMI action makes it possible to withdraw a caller from a queue back to the dialplan. The call will be signaled to leave the queue whenever it can, hence, it not guaranteed that the call will leave the queue. Optional custom data can be passed in the request, in the WithdrawInfo parameter. If the call successfully withdrawn the queue, it can be retrieved using the QUEUE_WITHDRAW_INFO variable. This can be useful for certain uses, such as dispatching the call to a specific extension. channel_internal_api ------------------ * CHANNEL(lastcontext) and CHANNEL(lastexten) are now available for use in the dialplan. res_pjsip_pubsub ------------------ * A new resource_list option, resource_display_name, indicates whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. * The Resource List Subscriptions (RLS) is dynamic now. The asterisk now updates current subscriptions to reflect the changes to the list on subscription refresh. If list items are added, removed, updated or do not exist anymore, the asterisk regenerates the resource list. ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.1.0 to Asterisk 19.2.0 ------------ ------------------------------------------------------------------------------ Applications ------------------ * added support for Danish syntax, playing the correct plural sound file dependen on where you have 1 or multipe messages based on the existing SE/NO code * added that we set DIALEDPEERNUMBER on the outgoing channels so it is avalible in b(content^extension^line) this add the same behaviour as Dial Core ------------------ * Bundled PJProject Build The build process has been updated to make pjproject troubleshooting and development easier. See third-party/pjproject/README-hacking.md or https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject for more info. ami ------------------ * An AMI event now exists for "Wink". app_mf ------------------ * Adds MF receiver and sender applications to support the R1 MF signaling protocol, including integration with the Dial application. app_queue ------------------ * added that we set DIALEDPEERNUMBER on the outgoing channels so it is avalible in b(content^extension^line) this add the same behaviour as Dial app_queues ------------------ * adding support for playing the correct en/et for nordic languages * Don't play sound_thanks if there is no leading hold_time message When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience" app_sendtext ------------------ * A ReceiveText application has been added that can be used in conjunction with the SendText application. app_voicemail ------------------ * added support for Danish syntax, playing the correct plural sound file dependen on where you have 1 or multipe messages based on the existing SE/NO code cdr ------------------ * A new CDR option, channeldefaultenabled, allows controlling whether CDR is enabled or disabled by default on newly created channels. The default behavior remains unchanged from previous versions of Asterisk (new channels will have CDR enabled, as long as CDR is enabled globally). chan_sip.c ------------------ * resolve issue with pickup on device that uses "183" and not "180" cli ------------------ * The "module refresh" command has been added, which allows unloading and then loading a module with a single command. func_json ------------------ * The JSON_DECODE dialplan function can now be used to parse JSON strings, such as in conjunction with CURL for using API responses. res_fax_spandsp ------------------ * Adds support for spandsp 3.0.0.
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