diff --git a/CHANGES b/CHANGES new file mode 100644 index 0000000000..401a886c62 --- /dev/null +++ b/CHANGES @@ -0,0 +1,8263 @@ +============================================================================== +=== +=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE +=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO +=== doc/CHANGES-staging/README.md FOR MORE DETAILS. +=== +=== This file documents the new and/or enhanced functionality added in +=== the Asterisk versions listed below. This file does NOT include +=== changes in behavior that would not be backwards compatible with +=== previous versions; for that information see the UPGRADE.txt file +=== and the other UPGRADE files for older releases. +=== +============================================================================== + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------ +------------------------------------------------------------------------------ + +app_broadcast +------------------ + * A Broadcast application is now available which allows + for asynchronous one-to-many and many-to-one channel audio. + +app_directory +------------------ + * A new option 's' has been added to the Directory() application that + will skip calling the extension and instead set the extension as + DIRECTORY_EXTEN channel variable. + +app_read +------------------ + * A new option 'e' has been added to allow Read() to return the + terminator as the dialed digits in the case where only the terminator + is entered. + +app_senddtmf +------------------ + * A new option has been added to SendDTMF() which will answer the + specified channel if it is not already up. If no channel is specified, + the current channel will be answered instead. + +app_signal +------------------ + * Adds Signal and WaitForSignal applications + which can be used for signaling or as a + simple message queue in the dialplan. + +func_json +------------------ + * Additional parsing capabilities have been added to the + JSON_DECODE function, including support for arrays + and recursive indexing. + +res_phoneprov +------------------ + * On multihomed Asterisk servers with dynamic SERVER template variables, + reloading this module is no longer required when re-provisioning your + phone to another interface address (e.g. when moving between VLANs.) + +res_pjsip_rfc3326 +------------------ + * Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in + addition to currently supported Q.850). The first header found will be used to set + the HANGUPCAUSE variable. + +res_pjsip_session +------------------ + * The overlap_context option now allows explicitly + specifying a context to use for overlap dialing matches. + +res_rtp_asterisk +------------------ + * This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is available 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 + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * The AOCMessage action can now be used to generate AOC-S messages. + +Add support for named capture agent. +------------------ + * A name for the capture agent can now be specified + using the capture_name option which, if specified, + will be sent to the HEP server. + +app_if +------------------ + * Adds the If, ElseIf, Else, EndIf, and ExitIf applications + for conditional execution of a block of code. + +app_mixmonitor +------------------ + * The d option for MixMonitor now allows deleting + the original recording when MixMonitor exits, + which can be useful when MixMonitor copies it + somewhere else before exiting. + + * Adds the c option to use the real Caller ID on + the channel in voicemail recordings as opposed + to the Connected Line. + +app_voicemail +------------------ + * The voicemail user option attachextrecs can + now be set to control whether external recordings + trigger voicemail email notifications. + +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. + +chan_dahdi +------------------ + * FXO channels (FXS signaled) that don't use callerid or + distinctive ring detection can now be configured + to enter the dialplan immediately using immediate=yes, + instead of waiting for at least one ring. + +pbx_builtins +------------------ + * It is now possible to not wait for media on + a channel when answering it using Answer, + by specifying the i option. + +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_aoc +------------------ + * Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages. + A new endpoint option, send_aoc, controls this. + +res_pjsip_header_funcs +------------------ + * The new PJSIP_HEADER_PARAM function now fully supports both + URI and header parameters. Both reading and writing + parameters are supported. + +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. + +xmldocs +------------------ + * The XML documentation can now be reloaded without restarting + Asterisk, which makes it possible to load new modules that + enforce documentation without restarting Asterisk. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------ +------------------------------------------------------------------------------ + +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 19.0.0 to Asterisk 20.0.0 ------------ +------------------------------------------------------------------------------ + +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). + +db +------------------ + * The DBPrefixGet AMI action now allows retrieving + all of the DB keys beginning with a particular + prefix. + +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 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. + + Added 4 built-in profiles: + "" + "" + "" + "" + 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. + +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 + +Channel-agnostic MF support +------------------ + * A SendMF application and PlayMF manager + application are now included to send + arbitrary standard R1 MF tones on the + current channel or another specified channel. + +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. + +Handle non-standard Meter metric type safely +------------------ + * A meter_support flag has been introduced that defaults to true to maintain current behaviour. + If disabled, a counter metric type will be used instead wherever a meter metric type was used, + the counter will have a "_meter" suffix appended to the metric name. + +MessageSend +------------------ + * The MessageSend AMI action has been updated to allow the Destination + and the To addresses to be provided separately. This brings the + MessageSend manager command in line with the capabilities of the + MessageSend dialplan application. + +ToneScan application +------------------ + * A new application, ToneScan, allows for + synchronous detection of call progress + signals such as dial tone, busy tone, + Special Information Tones, and modems. + +ami +------------------ + * An AMI event now exists for "Wink". + + * AMI events can now be globally disabled using + the disabledevents [general] setting. + +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_dtmfstore +------------------ + * New application which collects digits + dialed and stores them into + a specified variable. + +app_mf +------------------ + * Adds MF receiver and sender applications to support + the R1 MF signaling protocol, including integration + with the Dial application. + + * Adds an option to ReceiveMF to cap the + number of digits read at a user-specified + maximum. + +app_milliwatt +------------------ + * The Milliwatt application's existing behavior is + incorrect in that it plays a constant tone, which + is not how digital milliwatt test lines actually + work. + + An option is added so that a proper milliwatt test + tone can be provided, including a 1 second silent + interval every 10 seconds. However, for compatability + reasons, the default behavior remains unchanged. + +app_morsecode +------------------ + * Extends the Morsecode application by adding support for + American Morse code and adds a configurable option + for the frequency used in off intervals. + +app_originate +------------------ + * Codecs can now be specified for dialplan-originated + calls, as with call files and the manager action. + By default, only the slin codec is now used, instead + of all the slin* codecs. + +app_playback +------------------ + * A new option 'mix' is added to the Playback application that + will play by filename and say.conf. It will look on the format of the + name, if it is like say format it will play with say.conf if not it + will play the file name. + +app_queue +------------------ + * Reload behavior in app_queue has been changed so + queue and agent stats are not reset during full + app_queue module reloads. The queue reset stats + CLI command may still be used to reset stats while + Asterisk is running. + + * Add field to save the time value when a member enter a queue. + Shows this time in seconds using 'queue show' command and the + field LoginTime for responses for AMI the events. + + The output for the CLI command `queue show` is changed by added a + extra data field for the information of the time login time for each + member. + + * 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 + + * 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. + + * The m option now allows an override music on hold + class to be specified for the Queue application + within the dialplan. + +app_queue.c +------------------ + * Allow multiple files to be streamed for agent announcement. + +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_read +------------------ + * A new option allows the digit '#' to be read literally, + rather than used exclusively as the input terminator + character. + +app_sendtext +------------------ + * A ReceiveText application has been added that can be + used in conjunction with the SendText application. + +app_voicemail +------------------ + * Add a new 'S' option to VoiceMail which prevents the instructions + (vm-intro) from being played if a busy/unavailable/temporary greeting + from the voicemail user is played. This is similar to the existing 's' + option except that instructions will still be played if no user + greeting is available. + + * 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 + + * The r option has been added, which prevents deletion + of messages from VoiceMailMain, which can be + useful for shared mailboxes. + +apps +------------------ + * A new option 'mix' is added to the Playback application that + will play by filename and say.conf. It will look on the format of the + name, if it is like say format it will play with say.conf if not it + will play the file name. + +ari +------------------ + * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) + to ARI channel resources as 'protocol_id'. + + ASTERISK-30027 + +ast_coredumper +------------------ + * New options: + --pid= + Allows specification of an Asterisk instance when trying to + and the script can't determine it itself. + --libdir= + Allows specification of a non-standard installation directory + containing the Asterisk modules. + --(no-)rename + Renames the coredump and the output files with readable + timestamps. This is the default. + Removed unneeded or confusing options: + --append-coredumps + --conffile + --no-default-search + --tarball-uniqueid + Changed Variables: + COREDUMPS is now just "/tmp/core!(*.txt)" + DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ' + Changed behavior: + If you use 'running' or 'RUNNING' you no longer need to specify + '--no-default-search' to ignore existing coredumps. + +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_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. + + * A POLARITY function is now available that allows + getting or setting the polarity on a channel + from the dialplan. + +chan_iax2 +------------------ + * ANI2 (OLI) is now transmitted over IAX2 calls + as an information element. + + * Both a secret and an outkey may be specified at dial time, + since encryption is possible with RSA authentication. + +chan_pjsip +------------------ + * Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do. + + Add ability to read header by pattern using PJSIP_HEADER(). + + * 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 + +chan_sip.c +------------------ + * resolve issue with pickup on device that uses "183" and not "180" + +channel_internal_api +------------------ + * CHANNEL(lastcontext) and CHANNEL(lastexten) + are now available for use in the dialplan. + +cli +------------------ + * The "module refresh" command has been added, + which allows unloading and then loading a + module with a single command. + + * 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_channel +------------------ + * Adds the CHANNEL_EXISTS function to check for the existence + of a channel by name or unique ID. + +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_env.c +------------------ + * Two new functions, DIRNAME and BASENAME, are now + included which allow users to obtain the directory + or the base filename of any file. + +func_evalexten +------------------ + * This adds the EVAL_EXTEN function which may be + used to evaluate data at dialplan extensions. + +func_framedrop +------------------ + * New function to selectively drop specified frames + in either direction on a channel. + +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. + +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. + +func_scramble +------------------ + * Adds an audio scrambler function that may be used to + distort voice audio on a channel as a privacy + enhancement. + +func_strings +------------------ + * A new STRBETWEEN function is now included which + allows a substring to be inserted between characters + in a string. This is particularly useful for transforming + dial strings, such as adding pauses between digits + for a string of digits that are sent to another channel. + +func_vmcount +------------------ + * Multiple mailboxes may now be specified instead of just one. + +logger +------------------ + * Added the ability to define custom log levels in logger.conf + and use them in the Log dialplan application. Also adds a + logger show levels CLI command. + +res_agi +------------------ + * Agi command 'exec' can now be enabled + to evaluate dialplan functions and variables + by setting the variable AGIEXECFULL to yes. + +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_fax_spandsp +------------------ + * Adds support for spandsp 3.0.0. + +res_geolocation +------------------ + * Added res_geolocation which creates the core capabilities + to manipulate Geolocation information on SIP INVITEs. + +res_parking +------------------ + * An m option to Park and ParkAndAnnounce now allows + specifying a music on hold class override. + +res_pjproject +------------------ + * In pjproject.conf you can now map pjproject log levels + to the Asterisk TRACE log level. The default mappings + have therefore changed so that only pjproject levels + 3 and 4 are mapped to DEBUG and 5 and 6 are now mapped + to TRACE. Previously 3, 4, 5, and 6 were all mapped to + DEBUG. + +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. + +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. + +res_pjsip_registrar +------------------ + * Adds new PJSIP AOR option remove_unavailable to either + remove unavailable contacts when a REGISTER exceeds + max_contacts when remove_existing is disabled, or + prioritize unavailable contacts over other existing + contacts when remove_existing is enabled. + +res_pjsip_t38 +------------------ + * In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the + fallback use of the transport's bind address solve problems sending + media on systems that cannot send ipv4 packets on ipv6 sockets, and + certain other situations. This change extends both of these behaviors + to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific + problems on these systems, introducing a new option + endpoint/t38_bind_udptl_to_media_address. + +res_rtp_asterisk +------------------ + * When the address of the STUN server (stunaddr) is a name resolved via DNS, the + stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL) + expires. This allows the STUN server to change its IP address without having to + reload the res_rtp_asterisk module. + +res_tonedetect +------------------ + * Arbitrary tone detection is now available through a + WaitForTone application (blocking) and a TONE_DETECT + function (non-blocking). + +say.c +------------------ + * Adds SAYFILES function to retrieve the file names that would + be played by corresponding Say applications, such as + SayDigits, SayAlpha, etc. + + Additionally adds SayMoney and SayOrdinal applications. + +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 18.0.0 to Asterisk 19.0.0 ------------ +------------------------------------------------------------------------------ + +AMI Flash event +------------------ + * Hook flash events are now exposed as AMI events. + +Add variable support to Originate +------------------ + * The Originate application now allows + variables to be set on the new channel + through a new option. + +Core +------------------ + * Added debug logging categories that allow a user to output debug information + based on a specified category. This lets the user limit, and filter debug + output to data relevant to a particular context, or topic. For instance the + following categories are now available for debug logging purposes: + + dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet + + These debug categories can be enable/disable via an Asterisk CLI command: + + core set debug category [:] [category[: [] ...] + + If no sub-level is associated all debug statements for a given category are + output. If a sub-level is given then only those statements assigned a value + at or below the associated sub-level are output. + + * The location where the media cache stores its temporary files + is no longer hardcoded to /tmp but can now be configured separately + via the astcachedir config variable in asterisk.conf. + + The default location for astcachedir is now /var/cache/asterisk + instead of /tmp, please make sure to manually cleanup and/or + migrate the temporary files in /tmp after upgrading. + +MessageSend +------------------ + * The MessageSend dialplan application now takes an + optional third argument that can set the message's + "To" field on outgoing messages. It's an alternative + to using the MESSAGE(to) dialplan function. + + To prevent confusion with the first argument, currently + named "to", it's been renamed to "destination". + Its function, creating the request URI, hasn't changed. + + The online documentation has also been enhanced to + explain the behavior. + + Despite the changes in this commit, there should be + no impact to current users of MessageSend. + +New ConfKick application +------------------ + * Adds a ConfKick() application, which allows + a specific channel, all users, or all non-admin + users to be kicked from a conference bridge. + +New Reload application +------------------ + * Adds an application to reload modules + +PlaybackFinished has a new error state +------------------ + * The PlaybackFinished event now has a new state "failed" + that is used when the sound file was not played due to an error. + Before the state on PlaybackFinished was always "done". + + In case of multiple sound files to be played, + the PlaybackFinished is sent only once in the end of the list, + even in case of error. + +WaitForCondition application +------------------ + * This application provides a way to halt + dialplan execution until a provided + condition evaluates to true. + +app_confbridge +------------------ + * app_confbridge now has the ability to force the estimated bitrate on an SFU + bridge. To use it, set a bridge profile's remb_behavior to "force" and + set remb_estimated_bitrate to a rate in bits per second. The + remb_estimated_bitrate parameter is ignored if remb_behavior is something + other than "force". + +app_confbridge answer supervision control +------------------ + * app_confbridge now provides a user option to prevent + answer supervision if the channel hasn't been + answered yet. To use it, set a user profile's + answer_channel option to no. + +app_dial announcement option +------------------ + * The A option for Dial now supports + playing audio to the caller as well + as the called party. + +app_mixmonitor +------------------ + * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and + MixMonitorMute when the channel monitoring is started, stopped and muted (or + unmuted) respectively. + +app_voicemail +------------------ + * The VoiceMail application can now be configured to send greetings and + instructions via early media and only answering the channel when it is + time for the caller to record their message. This behavior can be + activated by passing the new 'e' option to VoiceMail. + + * You can now customize the "beep" tone or omit it entirely. + +chan_iax2 +------------------ + * You can now specify a default "auth" method in the + [general] section of iax.conf + +chan_pjsip +------------------ + * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and + returns unsuccessful if it's used on a channel prior to answering. + +chan_pjsip, app_transfer +------------------ + * Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed, + transfers can pass a protocol specific error code. + Example, in SIP 3xx-6xx represent any SIP specific error received when + performing a REFER. + +func_math: Three new dialplan functions +------------------ + * Introduce three new functions, MIN, MAX, and ABS, which can be used to + obtain the minimum or maximum of up to two integers or absolute value. + +func_odbc +------------------ + * Introduce an ARGC variable for func_odbc functions, along with a minargs + per-function configuration option. + + minargs enables enforcing of minimum count of arguments to pass to + func_odbc, so if you're unconditionally using ARG1 through ARG4 then + this should be set to 4. func_odbc will generate an error in this case, + so for example + + [FOO] + minargs = 4 + + and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a + potentially leaked ARG4 from Gosub(). + + ARGC is needed if you're using optional argument, to verify whether or + not an argument has been passed, else it's possible to use a leaked ARGn + from Gosub (app_stack). So now you can safely do + ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. + +func_volume now can be read +------------------ + * The VOLUME function can now also be used + to read existing values previously set. + +logger +------------------ + * Added a new log formatter called "plain" that always prints + file, function and line number if available (even for verbose + messages) and never prints color control characters. Most + suitable for file output but can be used for other channels + as well. + + You use it in logger.conf like so: + debug => [plain]debug + console => [plain]error,warning,debug,notice,pjsip_history + messages => [plain]warning,error,verbose + + * The dateformat option in logger.conf will now control the remote + console (asterisk -r -T) timestamp format. Previously, dateformat only + controlled the formatting of the timestamp going to log files and the + main console (asterisk -c) but only for non-verbose messages. + + Internally, Asterisk does not send the logging timestamp with verbose + messages to console clients. It's up to the Asterisk remote consoles + to format verbose messages. Asterisk remote consoles previously did + not load dateformat from logger.conf. + + Previously there was a non-configurable and hard-coded "%b %e %T" + dateformat that would be used no matter what on all verbose console + messages printed on remote consoles. + + Example: + logger.conf + dateformat=%F %T.%3q + + # asterisk -rvvv -T + [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. + [Mar 19 09:55:43] -- Goto (dialExten,s,1) + + Given the following example configuration in logger.conf, Asterisk log + files and the console, will log verbose messages using the given + timestamp. Now ensuring that all remote console messages are logged + with the same dateformat as other log streams. + + --- + [general] + dateformat=%F %T.%3q + + [logfiles] + console => notice,warning,error,verbose + full => notice,warning,error,debug,verbose + --- + + Now we have a globally-defined dateformat that will be used + consistently across the Asterisk main console, remote consoles, and + log files. + + Now we have consistent logging: + + # asterisk -rvvv -T + [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. + [2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1) + +res_pjsip +------------------ + * PJSIP transports can now be partially reloaded safely. This allows the + local_net and external_* options to be updated without restarting Asterisk. + + * PJSIP endpoints can now be configured to skip authentication when + handling OPTIONS requests by setting the allow_unauthenticated_options + configuration property to 'yes.' + + * PJSIP support of registrations of endpoints in multidomain + scenarios, where the endpoint contains the domain info + in pjsip.conf. + +res_pjsip_dialog_info_body_generator +------------------ + * PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and + remote elements by iterating through ringing channels and inserting + that info into NOTIFY packet sent to the endpoint. + +res_pjsip_messaging +------------------ + * Implemented the new "to" parameter of the MessageSend() + dialplan application. This allows a user to specify + a complete SIP "To" header separate from the Request URI. + We now also accept a destination in the same format + as Dial()... PJSIP/number@endpoint + +res_rtp_asterisk +------------------ + * By default Asterisk reports the PJSIP version in all + STUN packets it sends. + + This behaviour may not be desired in a production + environment and can now be disabled by setting the + stun_software_attribute option to 'no' in rtp.conf. + +res_srtp +------------------ + * SRTP replay protection has been added to res_srtp and + a new configuration option "srtpreplayprotection" has + been added to the rtp.conf config file. For security + reasons, the default setting is "yes". Buggy clients + may not handle this correctly which could result in + no, or one way, audio and Asterisk error messages like + "replay check failed". + +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 18.0.0 -------------------------- +------------------------------------------------------------------------------ + +Core +------------------ + * The Streams API becomes the home for the core ACN capabilities. + These include... + + * Parsing and formatting of codec negotiation preferences. + * Resolving pending streams and topologies with those configured + using configured preferences. + * Utility functions for creating string representations of + streams, topologies, and negotiation preferences. + + For codec negotiation preferences: + * Added ast_stream_codec_prefs_parse() which takes a string + representation of codec negotiation preferences, which + may come from a pjsip endpoint for example, and populates + a ast_stream_codec_negotiation_prefs structure. + * Added ast_stream_codec_prefs_to_str() which does the reverse. + * Added many functions to parse individual parameter name + and value strings to their respective enum values, and the + reverse. + + For streams: + * Added ast_stream_create_resolved() which takes a "live" stream + and resolves it with a configured stream and the negotiation + preferences to create a new stream. + * Added ast_stream_to_str() which create a string representation + of a stream suitable for debug or display purposes. + + For topology: + * Added ast_stream_topology_create_resolved() which takes a "live" + topology and resolves it, stream by stream, with a configured + topology stream and the negotiation preferences to create a new + topology. + * Added ast_stream_topology_to_str() which create a string + representation of a topology suitable for debug or display + purposes. + * Renamed ast_format_caps_from_topology() to + ast_stream_topology_get_formats() to be more consistent with + the existing ast_stream_get_formats(). + + Additional changes: + * A new function ast_format_cap_append_names() appends the results + to the ast_str buffer instead of replacing buffer contents. + +app_bridgeaddchan +------------------ + * The BridgeAdd application now behaves more like the Bridge application. + The application now sets the BRIDGERESULT channel variable to indicate + what happened when the channel resumes in dialplan. This is instead of + hanging up the channel on failure conditions. + +res_pjsip +------------------ + * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref + have been added to res_pjsip endpoints that specify the preferred order + of codecs to use between those received/sent in an SDP offer and those + set in the endpoint configuration. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * You can now specify an optional 'Content-Type' as an argument for the Asterisk + SendText manager action. + +ARI +------------------ + * A new parameter 'inhibitConnectedLineUpdates' is now available in the + 'bridges.addChannel' call. This prevents the identity of the newly connected + channel from being presented to other bridge members. + +ARI Channels +------------------ + * The Channel resource has a new sub-resource "externalMedia". + This allows an application to create a channel for the sole purpose + of exchanging media with an external server. Once created, this + channel could be placed into a bridge with existing channels to + allow the external server to inject audio into the bridge or + receive audio from the bridge. + See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI + for more information. + +Core +------------------ + * H.265/HEVC is now a supported video codec and it can be used by + specifying "h265" in the allow line. + Please note however, that handling of the additional SDP parameters + described in RFC 7798 section 7.2 is not yet supported. + +Features +------------------ + * Adds support for AudioSocket, a very simple bidirectional audio streaming + protocol. There are both channel and application interfaces. + + A description of the protocol can be found on the referenced wiki page. A + short talk about the reasons and implementation can be found on YouTube at + the link provided. + + ARI support has also been added via the existing "externalMedia" ARI + functionality. The UUID is specified using the arbitrary "data" field. + + Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket + YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI + +Messaging +------------------ + * In order to reduce the amount of AMI and ARI events generated, + the global "Message/ast_msg_queue" channel can be set to suppress + it's normal channel housekeeping events such as "Newexten", + "VarSet", etc. This can greatly reduce load on the manager + and ARI applications when the Digium Phone Module for Asterisk + is in use. To enable, set "hide_messaging_ami_events" in + asterisk.conf to "yes" In Asterisk versions <18, the default + is "no" preserving existing behavior. Beginning with + Asterisk 18, the option will default to "yes". + +STIR/SHAKEN +------------------ + * STIR/SHAKEN support has been added to Asterisk. Configuration is done in + stir_shaken.conf. There is a sample configuration file to help you get + started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's + set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken + to yes on the endpoint configuration object. This will add an Identity + header on outgoing INVITEs, and check for an Identity header on incoming + INVITEs. This option has been added to Alembic as well. + + The information received on an incoming INVITE can be checked using the + STIR_SHAKEN dialplan function. There are two variations: + + STIR_SHAKEN(count) + STIR_SHAKEN(0, verify_result) + + The first variation will tell you how many STIR/SHAKEN results are on the + channel. The second fetches information for a specific result. The first + parameter is the index, followed by what information you want to retrieve. + The available options are 'verify_result', 'identity', and 'attestation'. + +app_chanisavail +------------------ + * The ChanIsAvail application now tolerates empty positions in the supplied + device list. Dialplan can now be simplified by not having to check for + empty positions in the device list. + +app_confbridge +------------------ + * A new bridge profile option, maximum_sample_rate, has been added which sets + a maximum sample rate that the bridge will be mixed at. This allows the bridge + to move below the maximum sample rate as needed but caps it at the maximum. + + * A new option, "text_messaging", has been added to the user profile + which allows control over whether text messaging is enabled or + disabled for a user. If enabled (the default) text messages + will be sent to the user. If disabled no text messages will be + sent to the user. + +app_dial +------------------ + * The Dial application now tolerates empty positions in the supplied + destination list. Dialplan can now be simplified by not having to check + for empty positions in the destination list. If there are no endpoints to + dial then DIALSTATUS is set to CHANUNAVAIL. + +app_mixmonitor +------------------ + * An option 'S' has been added to MixMonitor. If used in combination with + the r() and/or t() options, if a frame is available to write to one of + those files but not the other, a frame of silence if written to the file + that does not have an audio frame. This should prevent the two files + from "drifting" when mixed after the fact. + + * If the 'filename' argument to MixMonitor() ended with '.wav49,' + Asterisk would silently convert the extension to '.WAV' when opening + the file for writing. This caused the MIXMONITOR_FILENAME variable to + reference the wrong file. The MIXMONITOR_FILENAME variable will now + reflect the name of the file that Asterisk actually used instead of + the filename that was passed to the application. + +app_page +------------------ + * The Page application now tolerates empty positions in the supplied + destination list. Dialplan can now be simplified by not having to check + for empty positions in the destination list. + +app_voicemail +------------------ + * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from + the Asterisk voicemail directory on startup. Some users that store their + voicemails on network storage devices experienced slow startup times due to the + relative expense of traversing the voicemail directory structure looking for + orphaned lock files. This feature has now been removed. + + Users who require the lock files to be removed at startup should modify their + startup scripts to do so before starting the asterisk process. + +chan_pjsip +------------------ + * A new dialplan function, PJSIP_MOH_PASSTHROUGH, has been added to chan_pjsip. This + allows the behaviour of the moh_passthrough endpoint option to be read or changed + in the dialplan. This allows control on a per-call basis. + +chan_rtp +------------------ + * The UnicastRTP channel driver provided by chan_rtp now accepts + ":" as an alternative to ":" in the destination. + The first AAAA (preferred) or A record resolved will be used as the destination. + The lookup is synchronous so beware of possible dialplan delays if you specify a + hostname. + +func_curl +------------------ + * A new parameter, httpheader, has been added to CURLOPT function. This parameter + allows to set custom http headers for subsequent calls off CURL function. + Any setting of headers will replace the default curl headers + (e.g. "Content-type: application/x-www-form-urlencoded") + + * A new option, followlocation, can now be enabled with the CURLOPT() + dialplan function. Setting this will instruct cURL to follow 3xx + redirects, which it does not by default. + +func_jitterbuffer +------------------ + * The JITTERBUFFER dialplan function now has an option to enable video synchronization + support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip) + the video is buffered according to the size of the audio jitterbuffer and is + synchronized to the audio. + +func_volume +------------------ + * Accept decimal number as argument. + +http +------------------ + * You can now disable the /httpstatus page served by Asterisk's built-in + HTTP server by setting 'enable_status' to 'no' in http.conf. + +minmemfree +------------------ + * The 'minmemfree' configuration option now counts memory allocated to + the filesystem cache as "free" because it is memory that is available + to the process. + +res_ari_channels +------------------ + * When creating a channel in ARI using the create call + you can now specify dialplan variables to be set as part + of the same operation. + +res_musiconhold +------------------ + * This fix allows a realtime moh class to be unregistered from the command + line. This is useful when the contents of a directory referenced by a + realtime moh class have changed. + The realtime moh class is then reloaded on the next request and uses the + new directory contents. + + * A new mode - playlist - has been added to res_musiconhold. This mode allows the + user to specify the files (or URLs) to play explicitly by putting them directly + in musiconhold.conf. + +res_pjsip +------------------ + * Added a new PJSIP system setting called disable_rport. + Default is no to keep support working as before. + + If it is false (default) it adds the 'rport' parameter in the outgoing request message. + If it is true it does not add the 'rport' parameter in the outgoing request message. + + This is a system option, but working as a global option. + +res_pjsip_endpoint_identifier_ip +------------------ + * In 'type = identify' sections, the addresses specified for the 'match' + clause can now include a port number. For IP addresses, the port is + provided by including a colon after the address, followed by the + desired port number. If supplied, the netmask should follow the port + number. To specify a port for IPv6 addresses, the address itself must + be enclosed in brackets to be parsed correctly. + +res_pjsip_logger +------------------ + * The PJSIP packet logger now has the following CLI commands: + + pjsip set logger pcap + + When used this will create a pcap file containing the incoming + and outgoing SIP packets, in unencrypted form. + + pjsip set logger console + + This allows you to toggle logging to console on and off. + + pjsip set logger host add + + This allows you to add an additional IP address or subnet + mask to logging, allowing you to log multiple instead of + just a single IP address or all traffic. + + The normal "pjsip set logger host" CLI command has also been + expanded to allow subnet masks as well. + +res_pjsip_session +------------------ + * When placing an outgoing call to a PJSIP endpoint the intent + of any requested formats will now be respected. If only an audio + format is requested (such as ulaw) but the underlying endpoint + does not support the format the resulting SDP will still only + contain an audio stream, and not any additional streams such as + video. + + * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref + have been added to res_pjsip endpoints that specify the preferred order + of codecs to use between those received/sent in an SDP offer and those + set in the endpoint configuration. + +res_rtp_asterisk +------------------ + * This change include a new cli command 'rtp show settings' + + The command display by general settings of rtp configuration. For this + point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum, + strictrtp, learning_min_sequential and icesupport. + + * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to + an ACL mechanism. + + As such six now options are now available: + + ice_deny + ice_permit + ice_acl + stun_deny + stun_permit + stun_acl + + These options have their obvious meanings as used elsewhere. + + Backwards compatibility was maintained by adding {stun,ice}_blacklist as + aliases for {stun,ice}_deny. + +res_sorcery_memory_cache +------------------ + * The SorceryMemoryCacheExpireObject AMI action and CLI + command allow expiring of a specific object within the + sorcery memory cache. This is done by removing the + object from the cache with the expectation that the + cache will then re-populate the object when it is next + needed. + + For full backend caching this does not occur. The cache + won't repopulate until an entire refresh is done resulting + in the possibility that objects are missing until that + time. + + The AMI action and CLI command will now not allow + expiring of an object if the cache is configured as a + full backend cache. Instead you must use either the + SorceryMemoryCacheExpire or SorceryMemoryCachePopulate + AMI actions or their associated CLI commands. + +taskprocessor.c +------------------ + * Added two new CLI commands to reset stats for taskprocessors. You can + reset stats for a single, specific taskprocessor ('core reset + taskprocessor '), or you can reset all taskprocessors + ('core reset taskprocessors'). These commands will reset the counter for + the number of tasks processed as well as the max queue size. + + * Added "like" support for 'core show taskprocessors'. Now you + can specify a specific set of taskprocessors (or just one) by + adding the keyword "like" to the above command, followed by + your search criteria. + +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 17.0.0 -------------------------- +------------------------------------------------------------------------------ + +Bridging +------------------ + * The bridging core no longer uses the stasis cache for bridge + snapshots. The latest bridge snapshot is now stored on the + ast_bridge structure itself. + + The following APIs are no longer available since the stasis cache + is no longer used: + ast_bridge_topic_cached() + ast_bridge_topic_all_cached() + + A topic pool is now used for individual bridge topics. + + The ast_bridge_cache() function was removed since there's no + longer a separate container of snapshots. + + A new function "ast_bridges()" was created to retrieve the + container of all bridges. Users formerly calling + ast_bridge_cache() can use the new function to iterate over + bridges and retrieve the latest snapshot directly from the + bridge. + + The ast_bridge_snapshot_get_latest() function was renamed to + ast_bridge_get_snapshot_by_uniqueid(). + + A new function "ast_bridge_get_snapshot()" was created to retrieve + the bridge snapshot directly from the bridge structure. + + The ast_bridge_topic_all() function now returns a normal topic + not a cached one so you can't use stasis cache functions on it + either. + + The ast_bridge_snapshot_type() stasis message now has the + ast_bridge_snapshot_update structure as it's data. It contains + the last snapshot and the new one. + +Channels +------------------ + * The core no longer uses the stasis cache for channels snapshots. + The following APIs are no longer available: + ast_channel_topic_cached() + ast_channel_topic_all_cached() + The ast_channel_cache_all() and ast_channel_cache_by_name() functions + now returns an ao2_container of ast_channel_snapshots rather than a + container of stasis_messages therefore you can't call stasis_cache + functions on it. + The ast_channel_topic_all() function now returns a normal topic, + not a cached one so you can't use stasis cache functions on it either. + The ast_channel_snapshot_type() stasis message now has the + ast_channel_snapshot_update structure as it's data. + ast_channel_snapshot_get_latest() still returns the latest snapshot. + +chan_sip +------------------ + * The chan_sip module is now deprecated, users should migrate to the + replacement module chan_pjsip. See guides at the Asterisk Wiki: + https://wiki.asterisk.org/wiki/x/tAHOAQ + https://wiki.asterisk.org/wiki/x/hYCLAQ + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------ +------------------------------------------------------------------------------ + +AttendedTransfer +------------------ + * A new application, this will queue up attended transfer to the given extension. + +BlindTransfer +------------------ + * A new application, this will redirect all channels currently + bridged to the caller channel to the specified destination. + +ConfBridge +------------------ + * Add "average_all", "highest_all", and "lowest_all" values for + the remb_behavior option. These values operate on a bridge + level instead of a per-source level. This means that a single + REMB value is calculated and sent to every sender, instead of + a REMB value that is unique for the specific sender.. + +Dial +------------------ + * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and + milliseconds between creation of the dialing channel and receiving the first + RINGING signal + + Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to + the PROGRESS signal. Shorter of these two times should be equivalent to + the PDD (Post Dial Delay) value + + Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution + versions of DIALEDTIME and ANSWEREDTIME + +RTP/ICE +------------------ + * You can now indicate that you'd like an ice_host_candidate's local address + to be published as well as the mapped address. See the sample rtp.conf + for more information. + +ReadExten +------------------ + * Add 'p' option to stop reading extension if user presses '#' key. + +pbx_dundi +------------------ + * The DUNDi PBX module now supports IPv4/IPv6 dual binding. + +res_pjsip +------------------ + * Added a new PJSIP global setting called norefersub. + Default is true to keep support working as before. + + res_pjsip_refer configures PJSIP norefersub capability accordingly. + + Checks the PJSIP global setting value. + If it is true (default) it adds the norefersub capability to PJSIP. + If it is false (disabled) it does not add the norefersub capability + to PJSIP. + + This is useful for Cisco switches that do not follow RFC4488. + +res_rtp_asterisk +------------------ + * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This + allows larger certificates to be used for the DTLS negotiation. By default this value + is 1200. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ---------- +------------------------------------------------------------------------------ + +ARI +------------------ + * Application event filtering is now supported. An application can now specify + an "allowed" and/or "disallowed" list(s) of event types. Only those types + indicated in the "allowed" list are sent to the application. Conversely, any + types defined in the "disallowed" list are not sent to the application. Note + that if a type is specified in both lists "disallowed" takes precedence. + + * A new REST API call has been added: 'move'. It follows the format + 'channels/{channelId}/move' and can be used to move channels from one application + to another without needing to exit back into the dialplan. An application must be + specified, but the passing a list of arguments to the new application is optional. + An example call would look like this: + + client.channels.move(channelId=chan.id, app='ari-example', appArgs='a,b,c') + + If the channel was inside of a bridge when switching applications, it will + remain there. If the application specified cannot be moved to, then the channel + will remain in the current application and an event will be triggered named + "ApplicationMoveFailed", which will provide the destination application's name + and the channel information. + +res_pjsip +------------------ + * A new configuration parameter "taskprocessor_overload_trigger" has been + added to the pjsip.conf "globals" section. The distributor currently stops + accepting new requests when any taskprocessor overload is triggered. The + new option allows you to completely disable overload detection (NOT + RECOMMENDED), keep the current behavior, or trigger only on pjsip + taskprocessor overloads. + +chan_pjsip +------------------ + * A new configuration parameter 'ignore_183_without_sdp' has been added + to the pjsip.conf "endpoints" section. If enabled, will make chan_pjsip + discard 183s that do not contain an SDP body, which can resolve no + ringback tone issues as well as making the behavior match chan_sip. + +MWI +------------------ + * A new module "res_mwi_devstate" has been added that allows subscriptions + to voicemail boxes using "presence" events. This allows common BLF keys + to act as voicemail waiting indicators. + +app_queue +------------------ + * Added the ability to set the wrapuptime per-member using the AddQueueMember + application. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------ +------------------------------------------------------------------------------ + +ARI +------------------ + * Whenever an ARI application is started, a context will be created for it + automatically as long as one does not already exist, following the format + 'stasis-'. Two extensions are also added to this context: a match-all + extension, and the 'h' extension. Any phone that registers under this context + will place all calls to the corresponding Stasis application. + +res_pjsip +------------------ + * Added "send_contact_status_on_update_registration" global configuration option + to enable sending AMI ContactStatus event when a device refreshes its registration. + +Core +------------------ + * Reworked the media indexer so it doesn't cache the index. Testing revealed + that the cache added no benefit but that it could consume excessive memory. + Two new index related functions were created: ast_sounds_get_index_for_file() + and ast_media_index_update_for_file() which restrict index updating to + specific sound files. The original ast_sounds_get_index() and + ast_media_index_update() calls are still available but since they no longer + cache the results internally, developers should re-use an index they may + already have instead of calling ast_sounds_get_index() repeatedly. If + information for only a single file is needed, ast_sounds_get_index_for_file() + should be called instead of ast_sounds_get_index(). + +Features +------------------ + * Before Asterisk 12, when using the automon or automixmon features defined + in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on + both channels, indicating the filename of the recording. + + When bridging was overhauled in Asterisk 12, the behavior was changed such + that the variable was only set on the peer channel and not on the channel + that initiated the automon or automixmon. + + The previous behavior has been restored so both channels receive the + channel variable when one of these features is invoked. + +app_voicemail +------------------ + * You can now specify a special context with the "aliasescontext" parameter + in voicemail.conf which will allow you to create aliases for physical + mailboxes. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------ +------------------------------------------------------------------------------ + +pbx_config +------------------ + * pbx_config will now find and process multiple 'globals' sections from + extensions.conf. Variables are processed in the order they are found + and duplicate variables overwrite the previous value. + +chan_pjsip +------------------ + * New dialplan function PJSIP_PARSE_URI added to parse an URI and return + a specified part of the URI. + +Core +------------------ + * ast_bt_get_symbols() now returns a vector of strings instead of an + array of strings. This must be freed with ast_bt_free_symbols. + +res_pjsip +------------------ + * New options 'trust_connected_line' and 'send_connected_line' have been + added to the endpoint. The option 'trust_connected_line' is to control + if connected line updates are accepted from this endpoint. + The option 'send_connected_line' is to control if connected line updates + can be sent to this endpoint. + The default value is 'yes' for both options. + +res_rtp_asterisk +------------------ + * The existing strictrtp option in rtp.conf has a new choice availabe, called + 'seqno', which behaves the same way as setting strictrtp to 'yes', but will + ignore the time interval during learning so that bursts of packets can still + trigger learning our source. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15 to Asterisk 16 -------------------- +------------------------------------------------------------------------------ + +app_fax +------------------ + * The app_fax module is now deprecated, users should migrate to the + replacement module res_fax. + +app_originate +------------------ + * An 'a' option has been added to the Originate dialplan application which + will execute the originate in an asynchronous fashion. If set then the + application will return immediately without waiting for the originated + channel to answer. + +Build System +------------------ + * MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built + with MALLOC_DEBUG can now successfully load binary modules built without + MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer + need to have a special build with it enabled. + + * Asterisk now depends on libjansson >= 2.11. If this version is not + available on your distro you can use `./configure --with-jansson-bundled`. + +app_macro +------------------ + * The app_macro module is now deprecated and by default it is no longer + built. Users should migrate to app_stack (Gosub). A warning is logged + the first time any Macro is used. + +app_setcallerid +------------------ + * The app_setcallerid module has been removed. The CALLERID dialplan function + should be used instead. + +chan_sip +------------------ + * New function SIP_HEADERS() enumerates all headers in the incoming INVITE. + + * The variable GET_TRANSFERRER_DATA set in the peer channel causes matching + headers be retrieved from the REFER message and made accessible to the + dialplan in the hash TRANSFER_DATA. + +chan_dahdi +------------------ + * Timeouts for reading digits from analog phones are now configurable in + chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout. + +AMI +------------------ + * The ContactStatus and Status fields for the manager events ContactStatus + and ContactStatusDetail are now set to "NonQualified" when a contact exists + but has not been qualified. + + * The "Newexten" event is now part of the "dialplan" class. The documentation + for Asterisk 15 already specified this, but the implementation was actually + using the "call" class instead. + +ARI +------------------ + * The ContactInfo event's contact_status field is now set to "NonQualified" + when a contact exists but has not been qualified. + +app_queue +------------------ + * Added the ability to set the wrapuptime in the configuration of member. + When set the wrapuptime on the member is used instead of the wrapuptime + defined for the queue itself. + + * Added predial handler support for caller and callee channels with the + B and b options respectively. This is similar to the predial support + in app_dial. + +res_config_sqlite +------------------ + * The res_config_sqlite module is now deprecated, users should migrate to the + replacement module res_config_sqlite3. + +res_monitor +------------------ + * The res_monitor module is now deprecated, users should migrate to the + replacement module app_mixmonitor. + +res_pjsip +------------------ + * A new AMI action, PJSIPShowAors, has been added which displays information + about all configured PJSIP AORs. + + * A new AMI action, PJSIPShowAuths, has been added which displays information + about all configured PJSIP Auths. + + * A new AMI action, PJSIPShowContacts, has been added which displays information + about all configured PJSIP Contacts. + +res_pjsip_registrar_expire +------------------ + * The res_pjsip_registrar_expire module has been removed. The functionality has + been moved into res_pjsip_registrar. + +func_audiohookinherit +------------------ + * The func_audiohookinherit module has been removed. Due to architectural changes + in Asterisk 12, audiohook inheritance is performed automatically and this + function now lacks function. + +cdr_syslog +------------------ + * The cdr_syslog module is now deprecated and by default it is no longer + built. + +cdr_sqlite +------------------ + * The cdr_sqlite module has been removed. Users should move to using the + cdr_sqlite3_custom module instead. + +format_jpeg +------------------ + * The format_jpeg module has been removed. + +pbx_dundi +------------------ + * DUNDi now supports IPv6 + +Core: +------------------ + * libedit is no longer available as an embedded library and must be provided + by the system. + * The STATIC_BUILD functionality has been removed as it has not been maintained + and has not worked in quite some time. + * The module loader now enforces inter-module dependencies. This ensures that + a module is not started before another it depends on, even if preload is used. + If a dependency is not available or fails to startup this will block any + dependants from startup. + * Parts of the Asterisk core which can load configuration from realtime are now + built-in modules. It is no longer necessary to preload realtime drivers as + they are always initialized before the built-in modules. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------ +------------------------------------------------------------------------------ + +res_pjsip +------------------ + * A new option 'suppress_q850_reason_headers' has been added to the endpoint + object. Some devices can't accept multiple Reason headers and get confused + when both 'SIP' and 'Q.850' Reason headers are received. This option allows + the 'Q.850' Reason header to be suppressed. The default value is 'no'. + +res_pjsip_endpoint_identifier_ip +------------------ + * Added regex support to the identify section match_header option. You + specify a regex instead of an explicit string by surrounding the header + value with slashes: + match_header = SIPHeader: /regex/ + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------ +------------------------------------------------------------------------------ + +Core +------------------ + * Core bridging and, more specifically, bridge_softmix have been enhanced to + relay received frames of type TEXT or TEXT_DATA to all participants in a + softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to + take advantage of this so when res_pjsip_messaging receives an in-dialog + MESSAGE message from a user in a conference call, it's relayed to all + other participants in the call. + +app_sendtext +------------------ + * Support Enhanced Messaging. SendText now accepts new channel variables + that can be used to override the To and From display names and set the + Content-Type of a message. Since you can now set Content-Type, other + text/* content types are now valid. + +app_confbridge +------------------ + * ConfbridgeList now shows talking status. This utilizes the same voice + detection as the ConfbridgeTalking event, so bridges must be configured + with "talk_detection_events=yes" for this flag to have meaning. + + * ConfBridge can now send events to participants via in-dialog MESSAGEs. + All current Confbridge events are supported, such as ConfbridgeJoin, + ConfbridgeLeave, etc. In addition to those events, a new event + ConfbridgeWelcome has been added that will send a list of all + current participants to a new participant. + +res_pjsip +------------------ + * Two new options have been added to the system and endpoint objects to + control whether, on outbound calls, Asterisk will accept updated SDP answers + during the initial INVITE transaction when 100rel is not in effect. + This usually happens when the INVITE is forked to multiple UASs and more + than one sends an SDP answer or when a single UAS needs to change a media + port to switch from custom ringback to the actual media destination. + + The 'follow_early_media_forked' option sets whether Asterisk will accept + the updated SDP when the To tag on the subsequent response is different than + that on the the previous response. This usually occurs in the forked INVITE + scenario. The default value is "yes" which is the current behavior. + + The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the + updated SDP when the To tag on the subsequent response is the same as that + on the previous response. This can occur when a UAS needs to switch media + ports from custom ringback to the final media path. The default value is + "no" which is the current behavior. + + These options have to be enabled system-wide in the system config section + of pjsip.conf as well as on individual endpoints that require the + functionality. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------ +------------------------------------------------------------------------------ + +Core +------------------ + * A new configuration option "genericplc_on_equal_codecs" was added to the + "plc" section of codecs.conf to allow generic packet loss concealment even + if no transcoding was originally needed. Transcoding via SLIN is forced + in this case. + +res_pjproject +------------------ + * Added the "cache_pools" option to pjproject.conf. Disabling the option + helps track down pool content mismanagement when using valgrind or + MALLOC_DEBUG. The cache gets in the way of determining if the pool contents + are used after free and who freed it. + +res_pjsip_notify +------------------ + * Extend the PJSIPNotify AMI command to send an in-dialog notify on a + channel. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------ +------------------------------------------------------------------------------ + +Core +------------------ + * During dialplan reload log messages are produced for each context, + extension and include. These messages are no longer printed by the + verbose loggers, they are now only logged as debug messages. + +app_confbridge +------------------ + * Added the Muted header to the ConfbridgeJoin AMI event to indicate the + participant's starting mute status. + + * Made the AMI ConfbridgeList action's ConfbridgeList events output all + the standard channel snapshot headers instead of a few hand-coded channel + snapshot headers. The benefit is that the CallerIDName gets disruptive + characters like CR, LF, Tab, and a few others escaped. However, an empty + CallerIDName is now output as "" instead of "". + +app_followme +------------------ + * Added a new prompt, connecting-prompt, which will be played + (if configured) to the "winner" callee before connecting the call. + +res_pjsip +------------------ + * Users who are matching endpoints by SIP header need to reevaluate their + global "endpoint_identifier_order" option in light of the "ip" endpoint + identifier method split into the "ip" and "header" endpoint identifier + methods. + + * The pjsip_transport_event feature introduced in 15.1.0 has been refactored. + Any external modules that may have used that feature (highly unlikely) will + need to be changed as the API has been altered slightly. + +res_pjsip_endpoint_identifier_ip +------------------ + * The endpoint identifier "ip" method previously recognized endpoints either + by IP address or a matching SIP header. The "ip" endpoint identifier method + is now split into the "ip" and "header" endpoint identifier methods. The + "ip" endpoint identifier method only matches by IP address and the "header" + endpoint identifier method only matches by SIP header. The split allows the + user to control the relative priority of the IP address and the SIP header + identification methods in the global "endpoint_identifier_order" option. + e.g., If you have two type=identify sections where one matches by IP address + for endpoint alice and the other matches by SIP header for endpoint bob then + you can now predict which endpoint is matched when a request comes in that + matches both. + +res_pjsip_pubsub +------------------ + * In an earlier release, inbound registrations on a reliable transport + were pruned on Asterisk restart since the TCP connection would have + been torn down and become unusable when Asterisk stopped. This same + process is now also applied to inbound subscriptions. Since this + required the addition of a new column to the ps_subscription_persistence + realtime table, users who store their subscriptions in a database will + need to run the "alembic upgrade head" process to add the column to + the schema. + +res_pjsip_transport_management +------------------ + * Since res_pjsip_transport_management provides several attack + mitigation features, its functionality moved to res_pjsip and + this module has been removed. This way the features will always + be available if res_pjsip is loaded. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------ +------------------------------------------------------------------------------ + +Core +------------------ + * Added the "cache_media_frames" option to asterisk.conf. Disabling the option + helps track down media frame mismanagement when using valgrind or + MALLOC_DEBUG. The cache gets in the way of determining if the frame is + used after free and who freed it. NOTE: This option has no effect when + Asterisk is compiled with the LOW_MEMORY compile time option enabled because + the cache code does not exist. + +chan_sip +------------------ + * Calls to invalid extensions are now reported as an ACL failure security event + "no_extension_match". + +res_rtp_asterisk +------------------ + * The X.509 certificate used for DTLS negotiation can now be automatically + generated. This is supported by res_pjsip by specifying + "dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you + would set "dtlsautogeneratecert = yes" either in the [general] section of + sip.conf or on a specific peer. + +res_pjsip +------------------ + * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint + being matched based only on IP address. To ensure no behavior change the + default has been changed to "username,ip". + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 15.0.0 to Asterisk 15.1.0 ------------ +------------------------------------------------------------------------------ + +res_pjsip +------------------ + * The "remove_existing" option now allows a registration to succeed by + displacing any existing contacts that now exceed the "max_contacts" count. + Any removed contacts are the next to expire. The behaviour change is + beneficial when "rewrite_contact" is enabled and "max_contacts" is greater + than one. The removed contact is likely the old contact created by + "rewrite_contact" that the device is refreshing. + +AMI +------------------ + * Added a new CancelAtxfer action that cancels an attended transfer. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14 to Asterisk 15 -------------------- +------------------------------------------------------------------------------ + +app_queue +------------------ + * PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has + been defined. + + * A new option, "announce-position-only-up," has been added that, when set to + yes, causes position announcements to only be played when the caller's + queue position has improved since the last time that we announced their + position. This default is no. + +Build System +------------------ + * '--with-pjproject-bundled' is now the default when running ./configure + It can be disabled with '--without-pjproject-bundled'. + + * A '--with-download-cache' option is now available which is equivalent to + setting '--with-sounds-cache' and '--with-externals-cache' to the same + value. The download cache can also be set via the AST_DOWNLOAD_CACHE + environment variable. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------ +------------------------------------------------------------------------------ + +res_pjsip +------------------ + * The "external_media_address" on transports is now resolved using dnsmgr and + when dnsmgr refreshes are enabled will be automatically updated with the new + IP address of a given hostname. + + * A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive + unsolicited MWI NOTIFY requests and make them available to other modules via + the stasis message bus. + +res_musiconhold +------------------ + * By default, when res_musiconhold reloads or unloads, it sends a HUP signal + to custom applications (and all descendants), waits 100ms, then sends a + TERM signal, waits 100ms, then finally sends a KILL signal. An application + which is interacting with an external device and/or spawns children of its + own may not be able to exit cleanly in the default times, expecially if sent + a KILL signal, or if it's children are getting signals directly from + res_musiconhoild. To allow extra time, the 'kill_escalation_delay' + class option can be used to set the number of milliseconds res_musiconhold + waits before escalating kill signals, with the default being the current + 100ms. To control to whom the signals are sent, the "kill_method" + class option can be set to "process_group" (the default, existing behavior), + which sends signals to the application and its descendants directly, or + "process" which sends signals only to the application itself. + + * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode + of a channel on a per-call basis. + +res_xmpp +----------------- + * OAuth 2.0 authentication is now supported when contacting Google. Follow the + instructions in xmpp.conf.sample to retrieve and configure the necessary + tokens. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------ +------------------------------------------------------------------------------ + +app_voicemail +------------------ + * A new global option "imap_poll_logout" was added to specify whether need to + disconnect from the IMAP server after polling of mailboxes. + Default: no + +res_pjsip +------------------ + * A new endpoint option "refer_blind_progress" was added to turn off notifying + the progress details on Blind Transfer. If this option is not set then + the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". + On default is enabled. + Some SIP phones like Mitel/Aastra or Snom keep the line busy until + receive "200 OK". + + * A new endpoint option "notify_early_inuse_ringing" was added to control + whether to notify dialog-info state 'early' or 'confirmed' on Ringing + when already INUSE. + + * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This + mode works similar to 'auto' except uses DTMF INFO as fallback instead of + INBAND. + +res_agi +------------------ + * The EAGI() application will now look for a dialplan variable named + EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that + EAGI provides. If not specified, it will continue to use the default signed + linear (slin). + +chan_pjsip +------------------ + * When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan + function any contact which is considered unreachable due to qualify being + enabled will no longer be called. + + * The asymmetric_rtp_codec option now also controls whether chan_pjsip will + send media as-is without transcoding if the codec has been negotiated in the + SDP. If set to "no" then Asterisk will only ever send the preferred codec + from the SDP, unless the remote side sends a different codec and we will + switch to match. + +Build System +------------------ + * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used + to pass arbitrary options to the bundled pjproject configure. + + * Automatically set the bundled pjproject configure --host and --build + options to match those supplied for the asterisk configure. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------ +------------------------------------------------------------------------------ + +res_rtp_asterisk +------------------ + * Added the stun_blacklist option to rtp.conf. Some multihomed servers have + IP interfaces that cannot reach the STUN server specified by stunaddr. + Blacklist those interface subnets from trying to send a STUN packet to find + the external IP address. Attempting to send the STUN packet needlessly + delays processing incoming and outgoing SIP INVITEs because we will wait + for a response that can never come until we give up on the response. + Multiple subnets may be listed. + +Logging +------------------- + * Added logger_queue_limit to the configuration options. + All log messages go to a queue serviced by a single thread + which does all the IO. This setting controls how big that + queue can get (and therefore how much memory is allocated) + before new messages are discarded. + The default is 1000. + +res_pjsip_config_wizard +------------------ + * Two new parameters have been added to the pjsip config wizard. + Setting 'sends_line_with_registrations' to true will cause the wizard + to skip the creation of an identify object to match incoming requests + to the endpoint and instead add the line and endpoint parameters to + the outbound registration object. + Setting 'outbound_proxy' is a shortcut for adding individual + endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy + parameters. + +res_hep_rtcp +------------------ + * If the 'call-id' value is specified for the uuid_type option and a + chan_sip channel is used the resulting HEP traffic will now contain the + SIP Call-ID instead of the Asterisk channel name. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------ +------------------------------------------------------------------------------ + +Build System +------------------ + * LOW_MEMORY no longer has an effect on Asterisk ABI. Symbols that were + previously suppressed by LOW_MEMORY are now replaced by stub functions. + Asterisk built with LOW_MEMORY can now successfully load binary modules + built without LOW_MEMORY and vice versa. + + * RADIUS backends for CEL and CDR can now also be built using the radcli + client library, in addition to the existing support for building them + using either freeradius or radiusclient-ng. + +Core +------------------ + * ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources + which use mtx_prof must now manually declare and initialize the variable. + +chan_sip +------------------ + * If an offer is received with optional SRTP (a media stream with RTP/AVP but + which contains a crypto line) chan_sip will now accept it and enable SRTP. + If you would like to do optional SRTP on outbound you will need to create + a dialplan that dials with it enabled initially and if it fails fall back to + without. + +res_pjsip +------------------ + * Added endpoint configuration parameter "preferred_codec_only". + This allow asterisk response to a SIP invite with the single most + preferred codec rather than advertising all joint codec capabilities. + This limits the other side's codec choice to exactly what we prefer. + +cdr_radius +------------------ + * To fix a memory leak the syslog channel is now empty if it has not been set + and used by a syslog channel in the logger. + +cel_radius +------------------ + * To fix a memory leak the syslog channel is now empty if it has not been set + and used by a syslog channel in the logger. + +RTP +------------------ + * New setting "rtp_pt_dynamic = 35" in asterisk.conf: + Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32 + formats. To avoid the message "No Dynamic RTP mapping available", the range + was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However, + when you use more than 32 formats and calls are not accepted by a remote + implementation, please report this and go back to rtp_pt_dynamic = 96. + + * A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set + to "yes" RTP dynamic payload types are assigned dynamically per RTP instance. + When set to "no" RTP dynamic payload types are globally initialized to pre- + designated numbers and function similar to static payload types. + +app_originate +------------------ + * Added support to gosub predial routines on both original channel and on the + created channel using options parameter (like app_dial) B() and b(). This + allows for adding variables to newly created channel or, e.g. setting callerid. + +CLI Commands +------------------ + * 'dialplan show' output will now show [config_file:line_number] instead of + [registrar] when that information is available. Currently only extensions + registered by pbx_config when loading/reloading will use this format. + +app_queue +------------------ + * Add 'QueueUpdate' application which can be used to track outbound calls + using app_queue. + +pbx_spool +------------------ + * Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that + attempt-specific behavior is possible. This is a 1-based number that + simply increases by 1 for each attempt. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now + contains a new optional parameter, 'MatchHeader', mapping to the new + configuration option 'match_header' for the corresponding 'identify' object. + It should be noted that since 'match_header' takes in a key: value pair, the + event parameter will contain a ':' as well. + +app_record +------------------ + * Added new 'u' option to Record() application which prevents Asterisk from + truncating silence from the end of recorded files. + +res_pjsip_outbound_registration +------------------ + * Outbound registrations are now refreshed when res_stun_monitor detects + a network change event has happened. + The 'pjsip send (un)register' CLI commands were updated to accept '*all' + as an argument to operate on all registrations. + The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'. + +app_voicemail +------------------ + * The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and + 'vm-newuser' configuration options in voicemail.conf. + + * Added 'fromstring' field to the voicemail boxes. If set, it will override + the global 'fromstring' field on a per-mailbox basis. + +func_channel +------------------ + * Added CHANNEL(callid) to retrieve the call log tag associated with the + channel. e.g., [C-00000000] Dialplan now has access to the call log + search key associated with the channel so it can be saved in case there + is a problem with the call. + +res_pjsip +------------------ + * A new transport parameter 'symmetric_transport' has been added. + When a request from a dynamic contact comes in on a transport with this + option set to 'yes', the transport name will be saved and used for + subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's + saved as a contact uri parameter named 'x-ast-txp' and will display with + the contact uri in CLI, AMI, and ARI output. On the outgoing request, + if a transport wasn't explicitly set on the endpoint AND the request URI + is not a hostname, the saved transport will be used and the 'x-ast-txp' + parameter stripped from the outgoing packet. To facilitate recreation of + subscriptions on asterisk restart, a new column 'contact_uri' needed to be + added to the ps_subcsription_persistence table. Since new columns were + added to both transport and subscription_persistence, an alembic upgrade + should be run to bring the database tables up to date. + + * A new option, allow_overlap, has been added to endpoints which allows + overlap dialing functionality to be enabled or disabled. The option defaults + to enabled. + +res_pjsip_transport_websocket +------------------ + * Removed non-secure websocket support. Firefox and Chrome have not allowed + non-secure websockets for quite some time so this shouldn't be an issue + for people. Attempting to use a non-secure websocket may or may not work + when Asterisk attempts to send SIP requests to do something like initiate + call hangup. + +res_pjsip_endpoint_identifier_ip +------------------ + * A new option has been added to the 'identify' configuration object, + 'match_header'. The 'match_header' attribute should contain a SIP + header: value pair that, When set, will cause inbound requests that contain + the matching SIP header/value pair to be associated with the corresponding + endpoint. This option is cumulative with the 'match' option, so that if + either option matches the request, the request is associated with the + endpoint. + + In a future release, this module will be renamed to something more + appropriate, as it now matches inbound requests on more than just IP + address. + +res_rtp_asterisk +----------------- + * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP + Data and Control Packets on a Single Port." So far, the only channel driver + that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on + a PJSIP endpoint in pjsip.conf to enable the feature. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------ +------------------------------------------------------------------------------ + +res_pjproject +------------------ + * Added new CLI command "pjproject set log level". The new command allows + the maximum PJPROJECT log levels to be adjusted dynamically and + independently from the set debug logging level like many other similar + module debug logging commands. + + * Added new companion CLI command "pjproject show log level" to allow the + user to see the current maximum pjproject logging level. + + * Added new pjproject.conf startup section "log_level' option to set the + initial maximum PJPROJECT logging level. + +res_pjsip_outbound_registration +------------------ + * Statsd no longer logs redundant status PJSIP.registrations.state changes + for internal state transitions that don't change the reported public status + state. + +res_pjsip_registrar +------------------ + * The PJSIPShowRegistrationInboundContactStatuses AMI command has been added + to return ContactStatusDetail events as opposed to + PJSIPShowRegistrationsInbound which just a dumps every defined AOR. + +res_pjsip +------------------ + * Six existing contact fields have been added to the end of the + ContactStatusDetail AMI event: + ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and + QualifyTimeout. Existing fields have not been disturbed. + +res_pjsip_endpoint_identifier_ip +------------------ + * SRV lookups can now be done on provided hostnames to determine additional + source IP addresses for requests. This is configurable using the + "srv_lookups" option on the identify and defaults to "yes". + +ARI +------------------ + * The 'ari set debug' command has been enhanced to accept 'all' as an + application name. This allows dumping of all apps even if an app + hasn't registered yet. + + * 'ari set debug' now displays requests and responses as well as events. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.1.0 to Asterisk 14.2.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * Events that reference a bridge may now contain two new optional fields: + - 'BridgeVideoSourceMode': the video source mode for the bridge. + Can be one of 'none', 'talker', or 'single'. + - 'BridgeVideoSource': the unique ID of the channel that is the video + source in this bridge, if one exists. + + * A new event, BridgeVideoSourceUpdate, has been added with a class + authorization of CALL. The event is raised when the video source changes + in a multi-party mixing bridge. + +ARI +------------------ + * The bridges resource now exposes two new operations: + - POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a + multi-party mixing bridge + - DELETE /bridges/{bridgeId}/videoSource: Remove the set video source, + reverting to talk detection for the video source + + * The bridge model in any returned response or event now contains the following + optional fields: + - video_mode: the video source mode for the bridge. Can be one of 'none', + 'talker', or 'single'. + - video_source_id: the unique ID of the channel that is the video source + in this bridge, if one exists. + + * A new event, BridgeVideoSourceChanged, has been added for bridges. + Applications subscribed to a bridge will receive this event when the source + of video changes in a mixing bridge. + + * The ARI major version has been bumped. There are not any known breaking changes + in ARI. The major version has been bumped because otherwise we can end up with + overlapping version numbers between different Asterisk versions. Now each major + version of Asterisk will bring with it a change in the major version of ARI. + The ARI version in Asterisk 14 is now 2.0.0. + +res_pjsip +------------------ + * Automatic dual stack support is now implemented. Depending on DNS resolution + and the transport used for sending a message the SIP signaling and SDP will + be updated with the correct IP address and protocol version. This means that + the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The + res_pjsip_multihomed module has also been moved into core res_pjsip to ensure + that messages are updated with the correct address information in all cases. + +chan_pjsip +------------------ + * The default behavior for RTP codecs has been changed. The sending codec will + now match the receiving codec. This can be turned off and behavior reverted + to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this + option is set then the sending and received codec are allowed to differ. + +CLI Commands +------------------ + * Three new CLI commands have been added for ARI: + - ari show apps: + Displays a listing of all registered ARI applications. + - ari show app : + Display detailed information about a registered ARI application. + - ari set debug : + Enable/disable debugging of an ARI application. When debugged, verbose + information will be sent to the Asterisk CLI. + + +Queue +------------------ + * A new dialplan variable, ABANDONED, is set when the call is not answered + by an agent. + +res_ari +------------------ + * The configuration file ari.conf now supports a channelvars option, which + specifies a list of channel variables to include in each channel-oriented + ARI event. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ------------ +------------------------------------------------------------------------------ + +Build System +------------------ + * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and + codec_siren14 binary modules hosted at downloads.digium.com can now be + automatically downloaded and installed during the Asterisk install + process. If selected in menuselect, when 'make install' is run, the + script will check the downloads site for a new version and download + and install it if needed. The '--with-externals-cache' option to + ./configure can be used to specify a location to cache the latest + tarballs so they don't have to be re-downloaded for every install. + +app_voicemail +------------------ + * Added "tps_queue_high" and "tps_queue_low" options. + The options can modify the taskprocessor alert levels for this module. + Additional information can be found in the sample configuration file at + config/samples/voicemail.conf.sample. + +res_pjsip_mwi +------------------ + * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration + options to tune taskprocessor alert levels. + + * Added "mwi_disable_initial_unsolicited" global configuration option + to disable sending unsolicited MWI to all endpoints on startup. + Additional information can be found in the sample configuration file at + config/samples/pjsip.conf.sample. + +chan_pjsip +------------------ + * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When + invoked, a re-INVITE or UPDATE request will be sent immediately to the + endpoint underlying the channel. When used in combination with the existing + dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP + channel to be re-negotiated and updated after session set up. + +res_pjsip +------------------ + * A new endpoint configuration parameter 'contact_user' has been added which + when set will override the default user set on Contact headers in outgoing + requests. + + * If you are using a sorcery realtime backend to store global res_pjsip + options (ps_globals table) then you now have to do a res_pjsip reload for + changes to these options to take effect. If you are using pjsip.conf to + configure these options then you already had to do a reload after making + changes. + + * Added "ignore_uri_user_options" global configuration option for + compatibility with an ITSP that sends URI user field options. When enabled + the user field is truncated at the first semicolon. + Example: + URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone" + The user field is "1235557890;phone-context=national" + Which is truncated to this: "1235557890" + + Note: The caller-id and redirecting number strings obtained from incoming + SIP URI user fields are now always truncated at the first semicolon. + +res_rtp_asterisk +------------------ + * An option, ice_blacklist, has been added which allows certain subnets to be + excluded from local ICE candidates. + +app_confbridge +------------------ + * Some sounds played into the bridge are played asynchronously. This, for + instance, allows a channel to immediately exit the ConfBridge without having + to wait for a leave announcement to play. + +app_dial +------------------ + * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels + when another channel answers the call. The default of ANSWERED_ELSEWHERE + is unchanged. + +res_ari +------------------ + * ARI events will all now include a new field in the root of the JSON message, + 'asterisk_id'. This will be the unique ID for the Asterisk system + transmitting the event. The value can be overridden using the 'entityid' + setting in asterisk.conf. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13 to Asterisk 14 -------------------- +------------------------------------------------------------------------------ + +AMI +----------------- + * A new event, "DialState" has been added. This is similar to "DialBegin" and + "DialEnd" in that it tracks the state of a dialed call. The difference is that + this indicates some intermediate state change in the dial attempt, such as + "RINGING", "PROGRESS", or "PROCEEDING". + +ARI +----------------- + * A new ARI method has been added to the channels resource. "create" allows for + you to create a new channel and place that channel into a Stasis application. + This is similar to origination except that the specified channel is not + dialed. This allows for an application writer to create a channel, perform + manipulations on it, and then delay dialing the channel until later. + + * To complement the "create" method, a "dial" method has been added to the + channels resource in order to place a call to a created channel. + + * All operations that initiate playback of media on a resource now support + a list of media URIs. The list of URIs are played in the order they are + presented to the resource. A new event, "PlaybackContinuing", is raised when + a media URI finishes but before the next media URI starts. When a list is + played, the "Playback" model will contain the optional attribute + "next_media_uri", which specifies the next media URI in the list to be played + back to the resource. The "PlaybackFinished" event is raised when all media + URIs are done. + + * Stored recordings now allow for the media associated with a stored recording + to be retrieved. The new route, GET /recordings/stored/{name}/file, will + transmit the raw media file to the requester as binary. + + + * "Dial" events have been modified to not only be sent when dialing begins and ends. + They now are also sent for intermediate states, such as "RINGING", "PROGRESS", and + "PROCEEDING". + +Applications +------------------ + +BridgeAdd +------------------ + * A new application in Asterisk, this will join the calling channel + to an existing bridge containing the named channel prefix. + +ChanSpy +------------------ + * Added the 'l' option, which forces ChanSpy's audiohook to use a long queue + to store the audio frames. This option is useful if audio loss is + experienced when using ChanSpy, but may introduce some delay in the audio + feed on the listening channel. + +Codecs +------------------ + * Added format attribute negotiation for the iLBC audio codec. Format attribute + negotiation is provided by the res_format_attr_ilbc module. iLBC 20 is the + default now. Falls back to iLBC 30, when the remote party requests this. + +ConfBridge +------------------ + * Added the ability to pass options to MixMonitor when recording is used with + ConfBridge. This includes the addition of the following configuration + parameters for the 'bridge' object: + - record_file_timestamp: whether or not to append the start time to the + recorded file name + - record_options: the options to pass to the MixMonitor application + - record_command: a command to execute when recording is finished + Note that these options may also be with the CONFBRIDGE function. + +ControlPlayback +------------------ + * Remote files can now be retrieved and played back. See the Playback + dialplan application for more details. + +FollowMe +------------------ + * It is now possible to disable the prompt from a callee by setting + 'enable_callee_prompt = no' in followme.conf. + +Playback +------------------ + * Remote files can now be retrieved and played back via the Playback and other + media playback dialplan applications. This is done by directly providing + the URL to play to the dialplan application: + same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav) + Note that unlike 'normal' media files, the entire URI to the file must be + provided, including the file extension. Currently, on HTTP and HTTPS URI + schemes are supported. + +Queue +------------------- + * Added field ReasonPause on QueueMemberStatus if set when paused, the reason + the queue member was paused. + + * Added field LastPause on QueueMemberStatus for time when started the last + pause for a queue member. + + * Show the time when started the last pause for queue member on CLI for command + 'queue show'. + +SMS +------------------ + * Added the 'n' option, which prevents the SMS from being written to the log + file. This is needed for those countries with privacy laws that require + providers to not log SMS content. + + +Channel Drivers +------------------ + +chan_dahdi +------------------ + * The CALLERID(ani2) value for incoming calls is now populated in featdmf + signaling mode. The information was previously discarded. + + * Added the force_restart_unavailable_chans compatibility option. When + enabled it causes Asterisk to restart the ISDN B channel if an outgoing + call receives cause 44 (Requested channel not available). + +chan_iax2 +------------------ + * The iax.conf forcejitterbuffer option has been removed. It is now always + forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer + on a channel it will be on the channel. + + * A new configuration parameters, 'calltokenexpiration', has been added that + controls the duration before a call token expires. Default duration is 10 + seconds. Setting this to a higher value may help in lagged networks or those + experiencing high packet loss. + + * Plaintext auth mode is deprecated and removed from possible default modes. + +chan_rtp (was chan_multicast_rtp) +------------------ + * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp. + + * The format for dialing a unicast RTP channel is: + UnicastRTP/[/[]] + Where is something like '127.0.0.1:5060'. + Where are in standard Asterisk flag options format: + c() - Specify which codec/format to use such as 'ulaw'. + e() - Specify which RTP engine to use such as 'asterisk'. + + * New options were added for a multicast RTP channel. The format for + dialing a multicast RTP channel is: + MulticastRTP//[/[][/[]]] + Where can be either 'basic' or 'linksys'. + Where is something like '224.0.0.3:5060'. + Where is something like '127.0.0.1:5060'. + Where are in standard Asterisk flag options format: + c() - Specify which codec/format to use such as 'ulaw'. + i(
) - Specify the interface address from which multicast RTP + is sent. + l() - Set whether packets are looped back to the sender. The + enable value can be 0 to set looping to off and non-zero to set + looping on. + t() - Set the time-to-live (TTL) value for multicast packets. + +chan_sip +------------------ + * New 'rtpbindaddr' global setting. This allows a user to define which + ipaddress to bind the rtpengine to. For example, chan_sip might bind + to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10). + + * DTLS related configuration options can now be set at a general level. + Enabling DTLS support, though, requires enabling it at the user + or peer level. + + * Added the possibility to set the From: header through the the SIP dial + string (populating the fromuser/fromdomain fields), complementing the + [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192). + NOTE: This is again separated by an exclamation mark, so the To: header may + not contain one of those. + + * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now. + Previously Asterisk dropped calls only with UDP transports. However with + longer international calls via TCP, the SIP channel might break, because + all hops on the Internet route must stay online (have not a single power + outage, for example). Therefore with Session-Timers enabled (which are + enabled at default), you might see additional dropped calls. Consequently + please, consider to go for session-timers=refuse in your sip.conf. + +chan_pjsip +------------------ + * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter + to the request URI and From URI if the user is determined to be a phone + number. + + * New 'moh_passthrough' endpoint setting. This will pass hold and unhold + requests through using SIP re-invites with sendonly and sendrecv accordingly. + + * Added the pjsip.conf system type disable_tcp_switch option. The option + allows the user to disable switching from UDP to TCP transports described + by RFC 3261 section 18.1.1. + + * New 'line' and 'endpoint' options added on outbound registrations. This + allows some identifying information to be added to the Contact of the + outbound registration. If this information is present on messages received + from the remote server the message will automatically be associated with the + configured endpoint on the outbound registration. + + +Core +------------------ + * The core of Asterisk uses a message bus called "Stasis" to distribute + information to internal components. For performance reasons, the message + distribution was modified to make use of a thread pool instead of a + dedicated thread per consumer in certain cases. The initial settings for + the thread pool can now be configured in 'stasis.conf'. + + * A new core DNS API has been implemented which provides a common interface + for DNS functionality. Modules that use this functionality will require that + a DNS resolver module is loaded and available. + + * Modified processing of command-line options to first parse only what + is necessary to read asterisk.conf. Once asterisk.conf is fully loaded, + the remaining options are processed. The -X option now applies to + asterisk.conf only. To enable #exec for other config files you must + set execincludes=yes in asterisk.conf. Any other option set on the + command-line will now override the equivalent setting from asterisk.conf. + + * The TLS core in Asterisk now supports X.509 certificate subject alternative + names. This way one X.509 certificate can be used for hosts that can be + reached under multiple DNS names or for multiple hosts. + + * The Asterisk logging system now supports JSON structured logging. Log + channels specified in logger.conf or added dynamically via CLI commands now + support an optional specifier prior to their levels that determines their + formatting. To set a log channel to format its entries as JSON, a formatter + of '[json]' can be set, e.g., + full => [json]debug,verbose,notice,warning,error + + * The core now supports a 'media cache', which stores temporary media files + retrieved from external sources. CLI commands have been added to manipulate + and display the cached files, including: + - 'media cache show ' - show all cached media files, or details about + one particular cached media file + - 'media cache refresh ' - force a refresh of a particular media file + in the cache + - 'media cache delete ' - remove an item from the cache + - 'media cache create ' - retrieve a URI and store it in the cache + + * The ability for device state hints to be automatically created as a result of + device state changes now exists in the PBX. This functionality is referred to + as "autohints" and is configurable in extensions.conf by placing "autohints=yes" + in the context. If enabled a device state hint will be automatically created + with the name of the device. + +* If Asterisk is built with systemd support, and run under systemd, it will + notify systemd of its state using sd_notify. Use 'Type=notify' in + asterisk.service. + +Functions +------------------ + * The func_odbc global option "single_db_connection" default value has been + changed to 'no'. + + +Formats +------------------ + * New module format_ogg_speex added which supports Speex codec inside + Ogg containers (filename extension .spx). + + +CHANNEL +------------------ + * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for + the hold status of a channel. + +CURL +------------------ + * The CURL function now supports a write option, which will save the retrieved + file to a location on disk. As an example: + same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav) + will save 'foo.wav' to /tmp. + +DTMF Features +------------------ + * The transferdialattempts default value has been changed from 1 to 3. The + transferinvalidsound has been changed from "pbx-invalid" to + "privacy-incorrect". These were changed to make DTMF transfers be more + user-friendly by default. + + +Resources +------------------ + +res_http_media_cache +------------------ + * A backend for the core media cache, this module retrieves media files from + a remote HTTP(S) server and stores them in the core media cache for later + playback. + +res_musiconhold +------------------ + * Added sort=randstart to the sort options. It sorts the files by name and + then chooses the first file to play at random. + * Added preferchannelclass=no option to prefer the application-passed class + over the channel-set musicclass. This allows separate hold-music from + application (e.g. Queue or Dial) specified music. + +res_resolver_unbound +------------------ + * Added a res_resolver_unbound module which uses the libunbound resolver library + to perform DNS resolution. This module requires the libunbound library to be + installed in order to be used. + +res_pjsip +------------------ + * A new SIP resolver using the core DNS API has been implemented. This relies on + external SIP resolver support in PJSIP which is only available as of PJSIP + 2.4. If this support is unavailable the existing built-in PJSIP SIP resolver + will be used instead. The new SIP resolver provides NAPTR support, improved + SRV support, and AAAA record support. + +res_pjsip_info_empty +-------------------- + * A new module that can respond to empty Content-Type INFO packets during call. + Some SBCs will terminate a call if their empty INFO packets are not responded + to within a predefined time. + +res_pjsip_outbound_registration +------------------------------- +* A new 'fatal_retry_interval' option has been added to outbound registration. + When set (default is zero), and upon receiving a failure response to an + outbound registration, registration is retried at the given interval up to + 'max_retries'. + +res_pjsip_outbound_publish +------------------ + * Added a new multi_user option that when set to 'yes' allows a given configuration + to be used for multiple users. + + +CEL Backends +------------------ + +cel_pgsql +------------------ + * Added a new option, 'usegmtime', which causes timestamps in CEL events + to be logged in GMT. + + * Added support to set schema where located the table cel. This settings is + configurable for cel_pgsql via the 'schema' in configuration file + cel_pgsql.conf. + + +CDR Backends +------------------ + +cdr_adaptive_odbc +------------------ + * Added the ability to set the character to quote identifiers. This + allows adding the character at the start and end of table and column + names. This setting is configurable for cdr_adaptive_odbc via the + quoted_identifiers in configuration file cdr_adaptive_odbc.conf. + +cdr_odbc +------------------ + * Added a new configuration option, "newcdrcolumns", which enables use of the + post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. + +cdr_csv +------------------ + * Added a new configuration option, "newcdrcolumns", which enables use of the + post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ---------- +------------------------------------------------------------------------------ + +chan_dahdi +------------------ + * Added "faxdetect_timeout" option. + The option determines how many seconds into a call before faxdetect + is disabled for the call. Setting the value to zero disables the timeout. + +res_pjsip +------------------ + * Added "fax_detect_timeout" to endpoint. + The option determines how many seconds into a call before fax_detect + is disabled for the call. Setting the value to zero disables the timeout. + + * Added "subscribe_context" to endpoint. + If specified, incoming SUBSCRIBE requests will be searched for the matching + extension in the indicated context. If no "subscribe_context" is specified, + then the "context" setting is used. + +res_rtp_asterisk +------------------ + * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS). + Enabling PFS is attempted by default, and is dependent on the configuration + of the module using TLS. + - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not + specify a ECDHE cipher suite in sip.conf, for example: + dtlscipher=AES128-SHA + - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters + into the private key file, e.g., sip.conf dtlsprivatekey. For example: + openssl dhparam -out ./dh.pem 2048 + - Because clients expect the server to prefer PFS, and because OpenSSL sorts + its cipher suites by bit strength, see "openssl ciphers -v DEFAULT". + Consider re-ordering your cipher suites in the respective configuration + file. For example: + dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256 + which forces PFS and requires at least DTLS 1.2. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 ----------- +------------------------------------------------------------------------------ + +Core +------------------ + * A channel variable FORWARDERNAME is now set which indicates which channel + was responsible for a forwarding requests received on dial attempt. + +func_odbc +------------------ + * Added new global option "single_db_connection". + Enabling this option func_odbc will use a single database connection per DSN. + This option is enabled by default. + +res_fax +------------------ + * Added FAXMODE variable to let dialplan know what fax transport was used. + FAXMODE variable is set to either "audio" or "T38". + +res_pjsip +------------------ + * Added "via_addr", "via_port", "call_id" to contacts. + As res_pjsip_nat rewrites contact's address, only the last Via header + can contain the source address of registered endpoint. + Also Call-Id header may contain the source address of registered endpoint. + Added new fields ViaAddress,CallID to AMI event ContactStatus + + * Endpoint IP Access Controls + Added new configuration Endpoint options: + "acl" - list of IP ACL section names in acl.conf + "deny" - List of IP addresses to deny access from + "permit" - List of IP addresses to permit access from + "contact_acl" - List of Contact ACL section names in acl.conf + "contact_deny" - List of Contact header addresses to deny + "contact_permit" - List of Contact header addresses to permit + + * Added "reg_server" to contacts. + If the Asterisk system name is set in asterisk.conf, it will be stored + into the "reg_server" field in the ps_contacts table to facilitate + multi-server setups. + + * When starting Asterisk, received traffic will now be ignored until Asterisk + has loaded all modules and is fully booted. + +res_hep +------------------ + * Added a new option, 'uuid_type', that sets the preferred source of the Homer + correlation UUID. The valid options are: + - call-id: Use the PJSIP SIP Call-ID header value + - channel: Use the Asterisk channel name + The default value is 'call-id'. In the event that a HEP module cannot find a + valid value using the specified 'uuid_type', the module may fallback to a + more readily available source for the correlation UUID. + +res_odbc +------------------ + * A new option has been added, 'max_connections', which sets the maximum number + of concurrent connections to the database. This option defaults to 1 which + returns the behavior to that of Asterisk 13.7 and prior. + +app_confbridge +------------------ + * Added a bridge profile option called regcontext that allows you to + dynamically register the conference bridge name as an extension into + the specified context. This allows tracking down conferences on multi- + server installations via alternate means (DUNDI for example). By default + this feature is not used. + +Codecs +------------------ + * Added the associated format name to 'core show codecs'. + +res_ari_channels +------------------ + * Added 'formats' to channel create/originate to allow setting the allowed + formats for a channel when no originator channel is available. Especially + useful for Local channel creation where no other format information is + available. 'core show codecs' can now be used to look up suitable format + names. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------ +------------------------------------------------------------------------------ + +res_parking: + - The dynamic parking lot creation channel variables PARKINGDYNAMIC, + PARKINGDYNCONTEXT, PARKINGDYNEXTEN, and PARKINGDYNPOS are now looked + for in the parker's channel instead of the parked channel. This is only + of significance if the parker uses blind transfer or the DTMF one-step + parking feature. You need to use the double underscore '__' inheritance + for these variables. The indefinite inheritance is also recommended + for the PARKINGEXTEN variable. + +res_pjsip +------------------ + * Added new global option (disable_multi_domain) to pjsip. + Disabling Multi Domain can improve realtime performace by reducing + number of database requsts. + +chan_pjsip +------------------ + * Added 'pjsip show channelstats' CLI command. + +res_pjsip_outbound_publish +------------------ + * Added support for setting the transport used on outbound publish + using the transport configuration option. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------ +------------------------------------------------------------------------------ + +res_pjsip_caller_id +------------------ + * Per RFC3325, the 'From' header is now anonymized on outgoing calls when + caller id presentation is prohibited. + +res_pjsip_config_wizard +------------------ + * A new command (pjsip export config_wizard primitives) has been added that + will export all the pjsip objects it created to the console or a file + suitable for reuse in a pjsip.conf file. + +Build System +------------------ + * To help insure that Asterisk is compiled and run with the same known + version of pjproject, a new option (--with-pjproject-bundled) has been + added to ./configure. When specified, the version of pjproject specified + in third-party/versions.mak will be downloaded and configured. When you + make Asterisk, the build process will also automatically build pjproject + and Asterisk will be statically linked to it. Once a particular version + of pjproject is configured and built, it won't be configured or built + again unless you run a 'make distclean'. + + To facilitate testing, when 'make install' is run, the pjsua and pjsystest + utilities and the pjproject python bindings will be installed in + ASTDATADIR/third-party/pjproject. + + The default behavior remains building with the shared pjproject + installation, if any. + +app_confbridge +------------------ + * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. + + * Added Muted header to AMI ConfbridgeListRooms action response list events + to indicate the muted conference state. + + * Added Muted column to CLI "confbridge list" output to indicate the muted + conference state and made the locked column a yes/no value instead of a + locked/unlocked value. + +REDIRECTING(reason) +------------------ + * The REDIRECTING(reason) value is now treated consistently between + chan_sip and chan_pjsip. + + Both channel drivers match incoming reason values with values documented + by REDIRECTING(reason) and values documented by RFC5806 regardless of + whether they are quoted or not. RFC5806 values are mapped to the + equivalent REDIRECTING(reason) documented value and is set in + REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a + quoted string version ('"unconditional"') is converted to + REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal + with 'cfu' instead of any of the aliases. + + The incoming 480 response reason text supported by chan_sip checks for + known reason values and if not matched then puts quotes around the reason + string and assigns that to REDIRECTING(reason). + + Both channel drivers send outgoing known REDIRECTING(reason) values as the + unquoted RFC5806 equivalent. User custom values are either sent as is or + with added quotes if SIP doesn't allow a character within the value as + part of a RFC3261 Section 25.1 token. Note that there are still + limitations on what characters can be put in a custom user value. e.g., + embedding quotes in the middle of the reason string is just going to cause + you grief. + + * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. + e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the + 'cfu' value. + +res_pjproject +------------------ + * This module is the successor of res_pjsip_log_forwarder. As well as + handling the log forwarding (which now displays as 'pjproject:0' instead + of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI. + This displays the compiled-in options of the pjproject installation + Asterisk is currently running against. + + * Another feature of this module is the ability to map pjproject log levels + to Asterisk log levels, or to suppress the pjproject log messages + altogether. Many of the messages emitted by pjproject itself are the result + of errors which Asterisk will ultimately handle so the messages can be + misleading or just noise. A new config file (pjproject.conf) has been added + to configure the mapping and a new CLI command (pjproject show log mappings) + has been added to display the mappings currently in use. + +res_pjsip +------------------ + * Transports are now reloadable. In testing, no in-progress calls were + disrupted if the ip address or port weren't changed, but the possibility + still exists. To make sure there are no unintentional drops, a new option + 'allow_reload', which defaults to 'no' has been added to transport. If + left at the default, changes to the particular transport will be ignored. + If set to 'yes', changes (if any) will be applied. + + * Added new global option (regcontext) to pjsip. When set, Asterisk will + dynamically create and destroy a NoOp priority 1 extension + for a given endpoint who registers or unregisters with us. + + * Endpoints and aors can now be identified by the username and realm in an + incoming Authorization header. To use this feature, add "auth_username" + to your endpoint's "identify_by" list. You can combine "auth_username" + and the original "username" to test both the From/To and Authorization + headers. For endpoints, the order is controlled by the global + "endpoint_identifier_order" setting. For matching aors to an endpoint + for inbound registration, the order is controlled by this option. + + * In conjunction with the "auth_username" change, 3 new options have been + added to the global configuration object that control how many unidentified + requests over a certain period from the same IP address can be received + before a security alert is generated. A new CLI command + "pjsip show unidentified_requests" will list the current candidates. + +res_pjsip_history +------------------ + * A new module, res_pjsip_history, has been added that provides SIP history + viewing/filtering from the CLI. The module is intended to be used on systems + with busy SIP traffic, where existing forms of viewing SIP messages - such + as the res_pjsip_logger - may be inadequate. The module provides two new + CLI commands: + - 'pjsip set history {on|off|clear}' - this enables/disables SIP history + capturing, as well as clears an existing history capture. Note that SIP + packets captured are stored in memory until cleared. As a result, the + history capture should only be used for debugging/viewing purposes, and + should *NOT* be left permanently enabled on a system. + - 'pjsip show history' - displays the captured SIP history. When invoked + with no options, the entire captured history is displayed. Two options + are available: + -- 'entry ' - display a detailed view of a single SIP message in + the history + -- 'where ...' - filter the history based on some expression. For more + information on filtering, view the current CLI help for the + 'pjsip show history' command. + +Voicemail +------------------ + * app_voicemail and res_mwi_external can now be built together. The default + remains to build app_voicemail and not res_mwi_external but if they are + both built, the load order will cause res_mwi_external to load first and + app_voicemail will be skipped. Use 'preload=app_voicemail.so' in + modules.conf to force app_voicemail to be the voicemail provider. + +res_pjsip_sdp_rtp +------------------ + * A new option (bind_rtp_to_media_address) has been added to endpoint which + will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the + media_address as well as using it in the SDP. If set, RTP packets will now + originate from the media address instead of the operating system's "primary" + ip address. + +res_rtp_asterisk +------------------ + * A new configuration section - ice_host_candidates - has been added to + rtp.conf, allowing automatically discovered ICE host candidates to be + overriden. This allows an Asterisk server behind a 1:1 NAT to send its + external IP as a host candidate rather than relying on STUN to discover it. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------ +------------------------------------------------------------------------------ + +Codecs +------------------ + * Added format attribute negotiation for the VP8 video codec. Format attribute + negotiation is provided by the res_format_attr_vp8 module. + +ConfBridge +------------------ + * A new "timeout" user profile option has been added. This configures the number + of seconds that a participant may stay in the ConfBridge after joining. When + the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT + is set to "TIMEOUT" on the channel. + +chan_sip +------------------ + * The websockets_enabled option has been added to the general section of + sip.conf. The option is enabled by default to match the previous behavior. + The option should be disabled when using res_pjsip_transport_websockets to + ensure chan_sip will not conflict with PJSIP websockets. + +Dialplan Functions +------------------ + * The HOLD_INTERCEPT dialplan function now actually exists in the source tree. + While support for the events was added in Asterisk 13.4.0, the function + accidentally never made it in. That function is now present, and will cause + the 'hold' raised by a channel to be intercepted and converted into an + event instead. + +res_pjsip_outbound_registration +------------------------------- + * If res_statsd is loaded and a StatsD server is configured, basic statistics + regarding the state of outbound registrations will now be emitted. This + includes: + - A GAUGE statistic for the overall number of outbound registrations, i.e.: + PJSIP.registrations.count + - A GAUGE statistic for the overall number of outbound registrations in a + particular state, e.g.: + PJSIP.registrations.state.Registered + +res_pjsip +------------------ + * The ability to use "like" has been added to the pjsip list and show + CLI commands. For instance: CLI> pjsip list endpoints like abc + + * If res_statsd is loaded and a StatsD server is configured, basic statistics + regarding the state of PJSIP contacts will now be emitted. This includes: + - A GAUGE statistic for the overall number of contacts in a particular + state, e.g.: + PJSIP.contacts.states.Reachable + - A TIMER statistic for the RTT time for each qualified contact, e.g.: + PJSIP.contacts.alice@@127.0.0.1:5061.rtt + +res_sorcery_memory_cache +------------------------ + * A new caching strategy, full_backend_cache, has been added which caches + all stored objects in the backend. When enabled all objects will be + expired or go stale according to the configuration. As well when enabled + all retrieval operations will be performed against the cache instead of + the backend. + +func_callerid +------------------- + * CALLERID(pres) is now documented as a valid alternative to setting both + CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers, + like chan_sip, don't make a distinction between the two: they take the + least public value from name-pres and num-pres. By using CALLERID(pres) + for reading and writing, you touch the same combined value in the dialplan. + The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres), + REDIRECTING(to-pres) and REDIRECTING(from-pres). + +res_endpoint_stats +------------------- + * A new module that emits StatsD statistics regarding Asterisk endpoints. + This includes a total count of the number of endpoints, the count of the + number of endpoints in the technology agnostic state of the endpoint - + online or offline - as well as the number of channels associated with each + endpoint. These are recorded as three different GAUGE statistics: + - endpoints.count + - endpoints.state.{unknown|offline|online} + - endpoints.{tech}.{resource}.channels + + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------ +------------------------------------------------------------------------------ + +Dialplan Functions +------------------ + * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id' + extraction option when using with the 'pjsip' signalling option. It will + return the SIP Call-ID associated with the INVITE request that established + the PJSIP channel. + +ARI +------------------ + * Two new endpoint related events are now available: PeerStatusChange and + ContactStatusChange. In particular, these events are useful when subscribing + to all event sources, as they provide additional endpoint related + information beyond the addition/removal of channels from an endpoint. + + * Added the ability to subscribe to all ARI events in Asterisk, regardless + of whether the application 'controls' the resource. This is useful for + scenarios where an ARI application merely wants to observe the system, + as opposed to control it. There are two ways to accomplish this: + (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll', + has been added that, when present and True, will subscribe all + specified applications to all ARI event sources in Asterisk. + (2) Via the applications resource. An ARI client can, at any time, subscribe + to all resources in an event source merely by not providing an explicit + resource. For example, subscribing to an event source of 'channels:' + as opposed to 'channels:12345' will subscribe the application to all + channels. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * A new ContactStatus event has been added that reflects res_pjsip contact + lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown. + + * Added the Linkedid header to the common channel headers listed for each + channel in AMI events. + +ARI +------------------ + * A new feature has been added that enables the retrieval of modules and + module information through an HTTP request. Information on a single module + can be also be retrieved. Individual modules can be loaded to Asterisk, as + well as unloaded and reloaded. + +* A new resource has been added to the 'asterisk' resource, 'config/dynamic'. + This resource allows for push configuration of sorcery derived objects + within Asterisk. The resource supports creation, retrieval, updating, and + deletion. Sorcery derived objects that are manipulated by this resource + must have a sorcery wizard that supports the desired operations. + + * A new feature has been added that allows for the rotation of log channels + through HTTP requests. + + +res_pjsip +------------------ +* A new 'g726_non_standard' endpoint option has been added that, when set to + 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it + is AAL2 packed on the channel. + +* A new 'rtp_keepalive' endpoint option has been added. This option specifies + an interval, in seconds, at which we will send RTP comfort noise packets to + the endpoint. This functions identically to chan_sip's "rtpkeepalive" option. + +* New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added. + These options specify the amount of time, in seconds, that Asterisk will wait + before terminating the call due to lack of received RTP. These are identical + to chan_sip's rtptimeout and rtpholdtimeout options. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------ +------------------------------------------------------------------------------ + +chan_pjsip +------------------ + * New 'rpid_immediate' option to control if connected line update information + goes to the caller immediately or waits for another reason to send the + connected line information update. See the online option documentation for + more information. Defaults to 'no' as setting it to 'yes' can result in + many unnecessary messages being sent to the caller. + + * The configuration setting 'progressinband' now defaults to 'no', which + matches the actual behavior of previous versions. + +res_pjsip +------------------ + * A new CLI command has been added: "pjsip show settings", which shows + both the global and system configuration settings. + + * A new aor option has been added: "qualify_timeout", which sets the timeout + in seconds for a qualify. The default is 3 seconds. This overrides the + hard coded 32 seconds in pjproject. + + * Endpoint status will now change to "Unreachable" when all contacts are + unavailable. When any contact becomes available, the endpoint will status + will change back to "Reachable". + + * A new global option has been added: "max_initial_qualify_time", which + sets the maximum amount of time from startup that qualifies should be + attempted on all contacts. + +res_ari_channels +------------------ + * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the + events data model. These events are raised when a channel indicates a hold + or unhold, respectively. + +func_holdintercept +------------------ + * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when + placed on a channel, intercepts hold/unhold indications signalled by the + channel and prevents them from moving on to other channels in a bridge with + the hold initiator. Instead, AMI or ARI events are raised indicating that + the channel wanted to place someone on hold. This allows external + applications to implement their own custom hold/unhold logic. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------ +------------------------------------------------------------------------------ + +chan_pjsip/app_transfer +------------------ + * The Transfer application, when used with chan_pjsip, now supports using + a PJSIP endpoint as the transfer destination. This is in addition to + explicitly specifying a SIP URI to transfer to. + +res_ari_channels +------------------ + * The ARI /channels resource now supports a new operation, 'redirect'. The + redirect operation will perform a technology and state specific redirection + on the channel to a specified endpoint or destination. In the case of SIP + technologies, this is either a 302 Redirect response to an on-going INVITE + dialog or a SIP REFER request. + +res_pjsip +------------------ + * A new 'endpoint_identifier_order' option has been added that allows one to + set the order by which endpoint identifiers are processed and checked. This + option is specified under the 'global' type configuration section. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------ +------------------------------------------------------------------------------ + + * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which + allow examining PJSIP AORs or contacts from the dialplan. + +res_pjsip_outbound_registration +------------------ + * The 'pjsip send unregister' command now stops further registrations. + + * A new command 'pjsip send register' has been added which allows you to + start or restart periodic registration. It can be used after a + 'send unregister' or after a 401 permanent error. + +res_pjsip_config_wizard +------------------ + * This is a new module that adds streamlined configuration capability for + chan_pjsip. It's targeted at users who have lots of basic configuration + scenarios like 'phone' or 'agent' or 'trunk'. Additional information + can be found in the sample configuration file at + config/samples/pjsip_wizard.conf.sample. + +res_fax +----------- + * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds + and is now configurable via the 't38timeout' configuration option in + res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'. + The default remains at 5000 milliseconds. + +PJSIP Transports +---------- + * The ca_list_path transport parameter has been added for TLS transports. This + option behaves similarly to the old sip.conf option "tlscapath". In order to + use this, you must be using PJProject version 2.4 or higher. + +ARI +------------------ + * The Originate operation now takes in an originator channel. The linked ID of + this originator channel is applied to the newly originated outgoing channel. + If using CEL this allows an association to be established between the two so + it can be recognized that the originator is dialing the originated channel. + + * "language" (the default spoken language for the channel) is now included in + the standard channel state output for suitable events. + + * The POST channels/{id} operation and the POST channels/{id}/continue operation + now have a new "label" parameter. This allows for origination or continuation + to a labeled priority in the dialplan instead of requiring a specific priority + number. The ARI version has been bumped to 1.7.0 as a result. + +AMI +------------------ + * "Language" (the default spoken language for the channel) is now included in + the standard channel state output for suitable events. + + * AMI actions that return a list of events have been made to return consistent + headers for the action response event starting the list and the list complete + event. The AMI version has been bumped to 2.7.0 as a result. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * Event NewConnectedLine is emitted when the connected line information on + a channel changes. + +ARI +------------------ + * Event ChannelConnectedLine is emitted when the connected line information + on a channel changes. + +Core Transfers +----------------- + +The features.conf general section has three new configurable options: + * transferdialattempts + * transferretrysound + * transferinvalidsound +For more information on what these options do, see the Asterisk wiki: + https://wiki.asterisk.org/wiki/x/W4fAAQ + +Channel Drivers +------------------ + +chan_pjsip +------------------ + * New 'media_encryption_optimistic' endpoint setting. This will use SRTP + when possible but does not consider lack of it a failure. + +res_pjsip_endpoint_identifer_ip +------------------ + * New CLI commands have been added: "pjsip show identif(y|ies)", which lists + all configured PJSIP identify objects + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12 to Asterisk 13 -------------------- +------------------------------------------------------------------------------ + +Overview +------------------ + +Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such, +the focus of development for this release of Asterisk was on improving the +usability and features developed in the previous Standard release, Asterisk 12. +Beyond a general refinement of end user features, development focussed heavily +on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk +REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the +new features include: + +* Asterisk security events are now provided via AMI, allowing end users to + monitor their Asterisk system in real time for security related issues. +* External control of Message Waiting Indicators (MWI) through both AMI and ARI. +* Reception/transmission of out of call text messages using any supported + channel driver/protocol stack through ARI. +* Resource List Server support in the PJSIP stack, providing subscriptions to + lists of resources and batched delivery of NOTIFY requests. +* Inter-Asterisk distributed device state and mailbox state using the PJSIP + stack. + +It is important to note that Asterisk 13 is built on the architecture developed +during the previous Standard release, Asterisk 12. Users upgrading to +Asterisk 13 should read about the new features in Asterisk 12 later in this file +(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the +UPGRADE-12.txt delivered with this release. In particular, users upgrading to +Asterisk 13 from a release prior to Asterisk 12 should read the specifications +on AMI, CDRs, and CEL on the Asterisk wiki: + * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ + * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ + * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ + +Many new featuers in Asterisk 13 were introduced in point releases of +Asterisk 12. Following this section - which documents the changes from all +versions of Asterisk 12 to Asterisk 13 - users should examine the new features +that were introduced in the point releases of Asterisk 12, as they are also +included in Asterisk 13. + +Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file +delivered with this release. + + +Build System +------------------ + * Sample config files have been moved from configs/ to a sub-folder of that + directory, samples. + + * The menuselect utility has been pulled into the Asterisk repository. As a + result, the libxml2 development library is now a required dependency for + Asterisk. + + * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference + counted objects will emit additional debug information to the refs log file + located in the standard Asterisk log file directory. This log file is useful + in tracking down object leaks and other reference counting issues. Prior to + this version, this option was only available by modifying the source code + directly. This change also includes a new script, refcounter.py, in the + contrib folder that will process the refs log file. Note that this replaces + the refcounter utility that could be built from the utils directory. + + +Applications +------------------ + +DahdiBarge +------------------ + * This module was deprecated and has been removed. Users of app_dahdibarge + should use ChanSpy instead. + +MixMonitor +------------------ + * New options to play a beep when starting a recording and stopping a recording + have been added. The option "p" will play a beep to the channel that starts + the recording. The option "P" will play a beep to the channel that stops the + recording. + +Queue +------------------ + * Queue rules can now be stored in a database table, queue_rules. Unlike other + RealTime tables, the queue_rules table is only examined on module load or + module reload. A new general setting has been added to queuerules.conf, + 'realtime_rules', which, when set to 'yes', will cause app_queue to look in + RealTime for additional queue rules to parse. Note that both the file and + the database can be used as a provide of queue rules when 'realtime_rules' + is set to 'yes'. + + When app_queue is reloaded, all rules are re-parsed and loaded into memory. + There is no caching of RealTime queue rules. + +ReadFile +------------------ + * This module was deprecated and has been removed. Users of app_readfile + should use func_env's FILE function instead. + +Say +------------------ + * The 'say' family of dialplan applications now support the Japanese + language. The 'language' parameter in say.conf now recognizes a setting of + 'ja', which will enable Japanese language specific mechanisms for playing + back numbers, dates, and other items. + * Counting, enumeration and dates now supports Icelandic grammar with the + 'language' parameter set to 'is'. + +SayCountPL +------------------ + * This module was deprecated and has been removed. Users of app_saycountpl + should use the Say family of applications. + +SetMusicOnHold +------------------ + * The SetMusicOnHold dialplan application was deprecated and has been removed. + Users of the application should use the CHANNEL function's musicclass + setting instead. + +WaitMusicOnHold +------------------ + * The WaitMusicOnHold dialplan application was deprecated and has been + removed. Users of the application should use MusicOnHold with a duration + parameter instead. + +VoiceMail +------------------ + * VoiceMail and VoiceMailMain now support the Japanese language. The + 'language' parameter in voicemail.conf now recognizes a setting of 'ja', + which will enable prompts to be played back using a Japanese grammatical + structure. Additional prompts are necessary for this functionality, + including: + - jb-arimasu: there is + - jb-arimasen: there is not + - jb-oshitekudasai: please press + - jb-ni: article ni + - jb-ga: article ga + - jb-wa: article wa + - jb-wo: article wo + + * Add the ability to specify multiple email addresses in configuration, + separated by a |. + + +CDR Backends +------------------ + +cdr_sqlite +----------------- + * This module was deprecated and has been removed. Users of cdr_sqlite + should use cdr_sqlite3_custom. + +cdr_pgsql +------------------ + * Added the ability to support PostgreSQL application_name on connections. + This allows PostgreSQL to display the configured name in the + pg_stat_activity view and CSV log entries. This setting is configurable + for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf. + + +CEL Backends +------------------ + +cel_pgsql +------------------ + * Added the ability to support PostgreSQL application_name on connections. + This allows PostgreSQL to display the configured name in the + pg_stat_activity view and CSV log entries. This setting is configurable + for cel_pgsql via the appname configuration setting in cel_pgsql.conf. + + +Channel Drivers +------------------ + +chan_dahdi +------------------ + * SS7 support now requires libss7 v2.0 or later. + + * Added SS7 support for connected line and redirecting. + + * Most SS7 CLI commands are reworked as well as new SS7 commands added. + See online CLI help. + + * Added several SS7 config option parameters described in + chan_dahdi.conf.sample. + +chan_gtalk +------------------ + * This module was deprecated and has been removed. Users of chan_gtalk + should use chan_motif. + +chan_h323 +------------------ + * This module was deprecated and has been removed. Users of chan_h323 + should use chan_ooh323. + +chan_jingle +------------------ + * This module was deprecated and has been removed. Users of chan_jingle + should use chan_motif. + +chan_pjsip +------------------ + * Added the CLI command 'pjsip list ciphers' so a user can know what + OpenSSL names are available on their system for the pjsip.conf cipher + option. + +chan_sip +------------------ + * The SIPPEER dialplan function no longer supports using a colon as a + delimiter for parameters. The parameters for the function should be + delimited using a comma. + + * The SIPCHANINFO dialplan function was deprecated and has been removed. Users + of the function should use the CHANNEL function instead. + + +Core +------------------ + +Account Codes +------------------ + * Added functional peeraccount support. Except for Queue, the + accountcode propagation is now consistently propagated to outgoing + channels before dialing. The channel accountcode can change from its + original non-empty value on channel creation for the following specific + reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an + originate method that can specify an accountcode value. Three, the + calling channel propagates its peeraccount or accountcode to the + outgoing channel's accountcode before dialing. The change has two + visible effects. One, local channels now cross accountcode and + peeraccount across the special bridge between the ;1 and ;2 channels + just like channels between normal bridges. Two, the + CHANNEL(peeraccount) value can now be set before Dial and FollowMe to + set the accountcode on the outgoing channel(s). + + For Queue, an outgoing channel's non-empty accountcode will not change + unless explicitly set by CHANNEL(accountcode). The change has three + visible effects. One, local channels now cross accountcode and + peeraccount across the special bridge between the ;1 and ;2 channels + just like channels between normal bridges. Two, the queue member will + get an accountcode if it doesn't have one and one is available from the + calling channel's peeraccount. Three, accountcode propagation includes + local channel members where the accountcodes are propagated early + enough to be available on the ;2 channel. + +AMI +------------------ + * New DeviceStateChanged and PresenceStateChanged AMI events have been added. + These events are emitted whenever a device state or presence state change + occurs. The events are controlled by res_manager_device_state.so and + res_manager_presence_state.so. If the high frequency of these events is + problematic for you, do not load these modules. + + * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They + work in basically the same way as the 'dialplan add extension' and + 'dialplan remove extension' CLI commands respectively. + + * New AMI action LoggerRotate reloads and rotates logger in the same manner + as CLI command 'logger rotate' + + * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the + functionality of CLI commands 'fax show sessions', 'fax show session', + and fax show stats' respectively. + + * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset + enable manager control over PRI debugging levels and file output. + + * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP + endpoint as long as a default outbound endpoint is set. This also applies + to the equivalent CLI command (pjsip send notify) + + * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections + that give information on Asterisk's attempts to qualify the endpoint. + + * The DialEnd event will now contain a Forward header if the dial is ending + due to the call being forwarded. The contents of the Forward header is the + extension in the number to which the call is being forwarded. + +CEL +------------------ + * The "bridge_technology" extra field key has been added to BRIDGE_ENTER + and BRIDGE_EXIT events. + +Features +------------------ + * Channel variables are now substituted in arguments passed to applications + run by using dynamic features. + +TLS +------------------ + * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS). + Enabling PFS is attempted by default, and is dependent on the configuration + of the module using TLS. + - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not + specify a ECDHE cipher suite in sip.conf, for example: + tlscipher=AES128-SHA:DES-CBC3-SHA + - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters + into the private key file, e.g., sip.conf tlsprivatekey. For example, the + default dh2048.pem - see + http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt + - Because clients expect the server to prefer PFS, and because OpenSSL sorts + its cipher suites by bit strength, see "openssl ciphers -v DEFAULT". + Consider re-ordering your cipher suites in the respective configuration + file. For example: + tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH + will use PFS when offered by the client. Clients which do not offer PFS + fall-back to AES-128 (or even 3DES, as recommended by RFC 3261). + + +Functions +------------------ + +JACK_HOOK +------------------ + * The JACK_HOOK function now supports audio with a sample rate higher than + 8kHz. + + +Resources +------------------ + +res_config_pgsql +------------------ + * Added the ability to support PostgreSQL application_name on connections. + This allows PostgreSQL to display the configured name in the + pg_stat_activity view and CSV log entries. This setting is configurable + for res_config_pgsql via the dbappname configuration setting in + res_pgsql.conf. + +res_pjsip_outbound_publish +------------------ + * A new module, res_pjsip_outbound_publish provides the mechanisms for sending + PUBLISH requests for specific event packages to another SIP User Agent. + +res_pjsip_pubsub +------------------ + * The publish/subscribe core module has been updated to support RFC 4662 + Resource Lists, allowing Asterisk to act as a Resource List Server (RLS). + Resource lists are configured in pjsip.conf under a new object type, + resource_list. Resource lists can contain either message-summary or presence + events, and can be composed of specific resources that provide the event or + other resource lists. + + * Inbound publication support is provided by a new object, inbound-publication. + This configures res_pjsip_pubsub to accept PUBLISH requests from a particular + resource. Which events are accepted is constructed dynamically; see + res_pjsip_publish_asterisk for more information. + +res_pjsip_publish_asterisk +------------------ + * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of + Asterisk information to other Asterisk servers. This module is intended only + for Asterisk to Asterisk exchanges of information. Currently, this includes + both mailbox state and device state information. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------ +------------------------------------------------------------------------------ + +ARI +------------------ + * Stored recordings now support a new operation, copy. This will take an + existing stored recording and copy it to a new location in the recordings + directory. + + * LiveRecording objects now have three additional fields that can be reported + in a RecordingFinished ARI event: + - total_duration: the duration of the recording + - talking_duration: optional. The duration of talking detected in the + recording. This is only available if max_silence_seconds was specified + when the recording was started. + - silence_duration: optional. The duration of silence detected in the + recording. This is only available if max_silence_seconds was specified + when the recording was started. + Note that all duration values are reported in seconds. + + * Users of ARI can now send and receive out of call text messages. Messages + can be sent directly to a particular endpoint, or can be sent to the + endpoints resource directly and inferred from the URI scheme. Text + messages are passed to ARI clients as TextMessageReceived events. ARI + clients can choose to receive text messages by subscribing to the particular + endpoint technology or endpoints that they are interested in. + + * The applications resource now supports subscriptions to all endpoints of + a particular channel technology. For example, subscribing to an eventSource + of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints. + +res_pjsip +------------------ + * The endpoint configuration object now supports 'accountcode'. Any channel + created for an endpoint with this setting will have its accountcode set + to the specified value. + +res_hep_rtcp +------------------ + * A new module, res_hep_rtcp, has been added that will forward RTCP call + statistics to a HEP capture server. See res_hep for more information. + +Functions +------------------ + * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now + unconditionally inherited through masquerades. As a side benefit, more + than one audiohook of a given type may persist through a masquerade now. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------ +------------------------------------------------------------------------------ + +AgentRequest +------------------ + * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to + connect with an incoming caller after being alerted to the presence + of the incoming caller. The most likely reason this would happen is + the agent did not acknowledge the call in time. + +AMI +------------------ + * New events have been added for the TALK_DETECT function. When the function + is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be + emitted to connected AMI clients indicating the start/stop of talking on + the channel. + +ARI +------------------ + * New event models have been aded for the TALK_DETECT function. When the + function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished + events will be emitted to connected WebSockets subscribed to the channel, + indicating the start/stop of talking on the channel. + +Functions +------------------ + * A new function, TALK_DETECT, has been added. When set on a channel, this + fucntion causes events indicating the starting/stoping of talking on said + channel to be emitted to both AMI and ARI clients. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------ +------------------------------------------------------------------------------ + +ARI +------------------ + * A new Playback URI 'tone' has been added. Tones are specified either as + an indication name (e.g. 'tone:busy') from indications.conf or as a tone + pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback + URIs in that they must be stopped manually and will continue to occupy + a channel's ARI control queue until they are stopped. They also can not + be rewound or fastforwarded. + + * User events can now be generated from ARI. Events can be signalled with + arbitrary json variables, and include one or more of channel, bridge, or + endpoint snapshots. An application must be specified which will receive + the event message (other applications can subscribe to it). The message + will also be delivered via AMI provided a channel is attached. Dialplan + generated user event messages are still transmitted via the channel, and + will only be received by a stasis application they are attached to or if + the channel is subscribed to. + +chan_sip +----------- + * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI + fields for prohibited callingpres information. Values are legacy, no, and + yes. By default, legacy is used. + trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When + dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI + headers are appended to outbound SIP messages just as they are with + allowed callingpres values, but data about the remote party's identity is + anonymized. + When sendrpid=rpid, only the remote party's domain is anonymized. + trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI + headers are not sent. + trust_id_outbound=yes - RPID/PAI headers are applied with the full remote + party information in tact even for prohibited callingpres information. + In the case of PAI, a Privacy: id header will be appended for prohibited + calling information to communicate that the private information should + not be relayed to untrusted parties. + +res_parking +------------------ + * Manager action 'Park' now takes an additional argument 'AnnounceChannel' + which can be used to announce the parked call's location to an arbitrary + channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two + parties in a one to one bridge, 'TimeoutChannel' is treated as having + parked 'Channel' like with the Park Call DTMF feature and will receive + announcements prior to being hung up. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------ +------------------------------------------------------------------------------ + +Record +------------------ + * Record application now has an option 'o' which allows 0 to act as an exit + key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF' + +ChanSpy +-------------------------- + * ChanSpy now accepts a channel uniqueid or a fully specified channel name + as the chanprefix parameter if the 'u' option is specified. + +ConfBridge +-------------------------- + * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic + conference user menus. + + * CONFBRIDGE dialplan function is now capable of removing dynamic conference + menus, bridge settings, and user settings that have been applied by the + CONFBRIDGE dialplan function. + + * The ConfBridge dialplan application now sets a channel variable, + CONFBRIDGE_RESULT, upon exiting. This variable can be used to determine + how a channel exited the conference. + + * Added conference user option 'announce_join_leave_review'. This option + implies 'announce_join_leave' with the added effect that the user will + be asked if they want to confirm or re-record the recording of their + name when entering the conference + +Directory +-------------------------- + * At exit, the Directory application now sets a channel variable + DIRECTORY_RESULT to one of the following based on the reason for exiting: + OPERATOR user requested operator by pressing '0' for operator + ASSISTANT user requested assistant by pressing '*' for assistant + TIMEOUT user pressed nothing and Directory stopped waiting + HANGUP user's channel hung up + SELECTED user selected a user from the directory and is routed + USEREXIT user pressed '#' from the selection prompt to exit + FAILED directory failed in a way that wasn't accounted for. Dang. + +Monitor +------------------ + * Monitor() - A new option, B(), has been added that will turn on a periodic + beep while the call is being recorded. + +MusicOnHold +-------------------------- + * MusicOnHold streams (all modes other than "files") now support wide band + audio too. + +Page +-------------------------- + * Added options 'b' and 'B' to apply predial handlers for outgoing calls + and for the channel executing Page respectively. + +PickupChan +-------------------------- + * PickupChan now accepts channel uniqueids of channels to pickup. + +Say +-------------------------- + * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set + to 'true' (case insensitive), then any Say application (SayNumber, + SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will + anticipate DTMF. If DTMF is received, these applications will behave like + the background application and jump to the received extension once a match + is established or after a short period of inactivity. + +MixMonitor +------------------------- + * A new function, MIXMONITOR, has been added to allow access to individual + instances of MixMonitor on a channel. + + * A new option, B(), has been added that will turn on a periodic beep while the + call is being recorded. + + +Channel Drivers +------------------------- + +chan_sip +------------------------- + * TEL URI support for inbound INVITE requests has been added. chan_sip will + now handle TEL schemes in the Request and From URIs. The phone-context in + the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on + the inbound channel. + +Core +------------------ + * Exposed sorcery-based configuration files like pjsip.conf to dialplans via + the new AST_SORCERY diaplan function. + + * Core Show Locks output now includes Thread/LWP ID if the platform + supports this feature. + + * New "logger add channel" and "logger remove channel" CLI commands have + been added to allow creation and deletion of dynamic logger channels + without configuration changes. These dynamic logger channels will only + exist until the next restart of asterisk. + +ARI +------------------ + * The live recording object on recording events now contains a target_uri + field which contains the URI of what is being recorded. + + * The bridge type used when creating a bridge is now a comma separated list of + bridge properties. Valid options are: mixing, holding, dtmf_events, and + proxy_media. + + * A channelId can now be provided when creating a channel, either in the + uri (POST channels/my-channel-id) or as query parameter. A local channel + will suffix the second channel id with ';2' unless provided as query + parameter otherChannelId. + + * A bridgeId can now be provided when creating a bridge, either in the uri + (POST bridges/my-bridge-id) or as a query parameter. + + * A playbackId can be provided when starting a playback, either in the uri + (POST channels/my-channel-id/play/my-playback-id / + POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter. + + * A snoop channel can be started with a snoopId, in the uri or query. + +AMI +------------------ + * Originate now takes optional parameters ChannelId and OtherChannelId, + used to set the UniqueId on creation. The other id is assigned to the + second channel when dialing LOCAL, or defaults to appending ;2 if only + the single Id is given. + + * The Mixmonitor action now has a "Command" header that can be used to + indicate a post-process command to run once recording finishes. + +RealTime +------------------ + * A new set of Alembic scripts has been added for CDR tables. This will create + a 'cdr' table with the default schema that Asterisk expects. + + +Functions +------------------ + * A new function was added: PERIODIC_HOOK. This allows running a periodic + dialplan hook on a channel. Any audio generated by this hook will be + injected into the call. + + +Resources +------------------ + +res_hep +------------------ + * A new module, res_hep, has been added, that acts as a generic packet + capture agent for the Homer Encapsulation Protocol (HEP) version 3. + It can be configured via hep.conf. Other modules can use res_hep to send + message traffic to a HEP capture server. + +res_hep_pjsip +------------------ + * A new module, res_hep_pjsip, has been added that will forward PJSIP + message traffic to a HEP capture server. See res_hep for more + information. + +res_pjsip +------------------ + * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now + be set as the named set of ToS values (cs0-cs7, af11-af43, ef). + + * Added the following new CLI commands: + - "pjsip show contacts" - list all current PJSIP contacts. + - "pjsip show contact" - show specific information about a current PJSIP + contact. + - "pjsip show channel" - show detailed information about a PJSIP channel. + +res_pjsip_multihomed +------------------ + * A new module, res_pjsip_multihomed handles situations where the system + Asterisk is running out has multiple interfaces. res_pjsip_multihomed + determines which interface should be used during message sending. + +res_pjsip_pidf_digium_body_supplement +------------------ + * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY + request body formatting for presence support in Digium phones. + +res_pjsip_send_to_voicemail +------------------ + * A new module, res_pjsip_send_to_voicemail allows for REFER requests with + particular headers to transfer a PJSIP channel directly to a particular + extension that has VoiceMail. This is intended to be used with Digium + phones that support this feature. + +res_pjsip_outbound_registration +------------------ + * A new CLI command has been added: "pjsip show registrations", which lists + all configured PJSIP registrations + + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * Added a new module that provides AMI control over MWI within Asterisk, + res_mwi_external_ami. Note that this module depends on res_mwi_external; + for more information on enabling this module, see res_mwi_external. + This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as + the MWIGet/MWIGetComplete events. + + * The DialStatus field in the DialEnd event can now contain additional + statuses that convey how the dial operation terminated. This includes + ABORT, CONTINUE, and GOTO. + + * AMI will now emit security events. A new class authorization has been + added in manager.conf for the security events, 'security'. The new events + are: + - FailedACL - raised when a request violates an ACL check + - InvalidAccountID - raised when a request fails an authentication + check due to an invalid account ID + - SessionLimit - raised when a request fails due to exceeding the + number of allowed concurrent sessions for a service + - MemoryLimit - raised when a request fails due to an internal memory + allocation failure + - LoadAverageLimit - raised when a request fails because a configured + load average limit has been reached + - RequestNotAllowed - raised when a request is not allowed by + the service + - AuthMethodNotAllowed - raised when a request used an authentication + method not allowed by the service + - RequestBadFormat - raised when a request is received with bad formatting + - SuccessfulAuth - raised when a request successfully authenticates + - UnexpectedAddress - raised when a request has a different source address + then what is expected for a session already in progress with a service + - ChallengeResponseFailed - raised when a request's attempt to authenticate + has been challenged, and the request failed the authentication challenge + - InvalidPassword - raised when a request provides an invalid password + during an authentication attempt + - ChallengeSent - raised when an Asterisk service send an authentication + challenge to a request + - InvalidTransport - raised when a request attempts to use a transport not + allowed by the Asterisk service + + * Bridge related events now have two additional fields: BridgeName and + BridgeCreator. BridgeName is a descriptive name for the bridge; + BridgeCreator is the name of the entity that created the bridge. This + affects the following events: ConfbridgeStart, ConfbridgeEnd, + ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord, + ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer, + AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave + +ARI +------------------ + * The Bridge data model now contains the additional fields 'name' and + 'creator'. The 'name' field conveys a descriptive name for the bridge; + the 'creator' field conveys the name of the entity that created the bridge. + This affects all responses to HTTP requests that return a Bridge data model + as well as all event derived data models that contain a Bridge data model. + The POST /bridges operation may now optionally specify a name to give to + the bridge being created. + + * Added a new ARI resource 'mailboxes' which allows the creation and + modification of mailboxes managed by external MWI. Modules res_mwi_external + and res_stasis_mailbox must be enabled to use this resource. For more + information on external MWI control, see res_mwi_external. + + * Added new events for externally initiated transfers. The event + BridgeBlindTransfer is now raised when a channel initiates a blind transfer + of a bridge in the ARI controlled application to the dialplan; the + BridgeAttendedTransfer event is raised when a channel initiates an + attended transfer of a bridge in the ARI controlled application to the + dialplan. + + * Channel variables may now be specified as a body parameter to the + POST /channels operation. The 'variables' key in the JSON is interpreted + as a sequence of key/value pairs that will be added to the created channel + as channel variables. Other parameters in the JSON body are treated as + query parameters of the same name. + +HTTP +------------------ + * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be + automatically handled by the HTTP server if a request is received with a + Transfer-Encoding type of "chunked". + +res_pjsip +------------------ + * Path support has been added with the 'support_path' option in registration + and aor sections. + + * A 'debug' option has been added to the globals section that will allow + sip messages to be logged. + + * A 'set_var' option has been added to endpoints that will automatically + set the desired variable(s) on a channel created for that endpoint. + + * Several new tables and columns have been added to the realtime schema for + the res_pjsip related modules. See the UPGRADE.txt notes for updating + the database schema. + +res_mwi_external +------------------ + * A new module, res_mwi_external, has been added to Asterisk. This module + acts as a base framework that other modules can build on top of to allow + an external system to control MWI within Asterisk. For implementations + that make use of res_mwi_external, see res_mwi_external_ami and + res_ari_mailboxes. Note that res_mwi_external conflicts with other modules + that may produce MWI themselves, such as app_voicemail. res_mwi_external + and other modules that depend on it cannot be built or loaded with + app_voicemail present. + +res_pjsip +------------------ + * DNS functionality will now automatically be enabled if the system configured + nameservers can be retrieved. If the system configured nameservers can not be + retrieved the functionality will resort to using system resolution. Functionality + such as SRV records and failover will not be available if system resolution + is in use. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 11 to Asterisk 12 -------------------- +------------------------------------------------------------------------------ + +Overview +------------------ + +Asterisk 12 is a standard release of the Asterisk project. As such, the +focus of development for this release was on core architectural changes and +major new features. This includes: + * A more flexible bridging core based on the Bridging API + * A new internal message bus, Stasis + * Major standardization and consistency improvements to AMI + * Addition of the Asterisk RESTful Interface (ARI) + * A new SIP channel driver, chan_pjsip +In addition, as the vast majority of bridging in Asterisk was migrated to the +Bridging API used by ConfBridge, major changes were made to most of the +interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL. + +Specifications have been written for the affected interfaces. These +specifications are available on the Asterisk wiki: + * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ + * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ + * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ + +It is *highly* recommended that anyone migrating to Asterisk 12 read the +information regarding its release both in this file and in the accompanying +UPGRADE.txt file. More detailed information on the major changes can be found +on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ. + + +Build System +------------------ + * Added build option DISABLE_INLINE. This option can be used to work around a + bug in gcc. For more information, see + http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816 + + * Removed the CHANNEL_TRACE development mode build option. Certain aspects of + the CHANNEL_TRACE build option were incompatible with the new bridging + architecture. + + * Asterisk now optionally uses libxslt to improve XML documentation generation + and maintainability. If libxslt is not available on the system, some XML + documentation will be incomplete. + + * Asterisk now depends on libjansson. If a package of libjansson is not + available on your distro, please see http://www.digip.org/jansson/. + + * Asterisk now depends on libuuid and, optionally, uriparser. It is + recommended that you install uriparser, even if it is optional. + + * The new SIP stack and channel driver uses a particular version of PJSIP. + Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on + configuring and installing PJSIP for usage with Asterisk. + + * Optional API was re-implemented to be more portable, and no longer requires + weak reference support from the compiler. The build option OPTIONAL_API may + be disabled to disable Optional API support. + +Applications +------------------ + +AgentLogin +------------------ + * Along with AgentRequest, this application has been modified to be a + replacement for chan_agent. The act of a channel calling the AgentLogin + application places the channel into a pool of agents that can be + requested by the AgentRequest application. Note that this application, as + well as all other agent related functionality, is now provided by the + app_agent_pool module. See chan_agent and AgentRequest for more information. + + * This application no longer performs agent authentication. If authentication + is desired, the dialplan needs to perform this function using the + Authenticate or VMAuthenticate application or through an AGI script before + running AgentLogin. + + * If this application is called and the agent is already logged in, the + dialplan will continue execution with the AGENT_STATUS channel variable set + to ALREADY_LOGGED_IN. + + * The agents.conf schema has changed. Rather than specifying agents on a + single line in comma delineated fashion, each agent is defined in a separate + context. This allows agents to use the power of context templates in their + definition. + + * A number of parameters from agents.conf have been removed. This includes + maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat, + urlprefix, and savecallsin. These options were obsoleted by the move from + a channel driver model to the bridging/application model provided by + app_agent_pool. + +AgentRequest +------------------ + * A new application, this will request a logged in agent from the pool and + bridge the requested channel with the channel calling this application. + Logged in agents are those channels that called the AgentLogin application. + If an agent cannot be requested from the pool, the AGENT_STATUS dialplan + application will be set with an appropriate error value. + +AgentMonitorOutgoing +------------------ + * This application has been removed. It was a holdover from when + AgentCallbackLogin was removed. + +AlarmReceiver +------------------ + * Added support for additional Ademco DTMF signalling formats, including + Express 4+1, Express 4+2, High Speed and Super Fast. + + * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum + call time, in milliseconds, to run the application. + + * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the + maximum number of times to retry the call. + + * Added a new configuration option answait. If set, the AlarmReceiver + application will wait the number of milliseconds specified by answait + after the channel has answered. Valid values range between 500 + milliseconds and 10000 milliseconds. + + * Added configuration option no_group_meta. If enabled, grouping of metadata + information in the AlarmReceiver log file will be skipped. + +Answer +------------------ + * It is now no longer possible to bypass updating the CDR on the channel + when answering. CDRs reflect the state of the channel and will always + reflect the time they were Answered. + +BridgeWait +------------------ + * A new application in Asterisk, this will place the calling channel + into a holding bridge, optionally entertaining them with some form of + media. Channels participating in a holding bridge do not interact with + other channels in the same holding bridge. Optionally, however, a channel + may join as an announcer. Any media passed from an announcer channel is + played to all channels in the holding bridge. Channels leave a holding + bridge either when an optional timer expires, or via the ChannelRedirect + application or AMI Redirect action. + +ConfBridge +------------------ + * All participants in a bridge can now be kicked out of a conference room + by specifying the channel parameter as 'all' in the ConfBridge kick CLI + command, i.e., 'confbridge kick all' + + * CLI output for the 'confbridge list' command has been improved. When + displaying information about a particular bridge, flags will now be shown + for the participating users indicating properties of that user. + + * The ConfbridgeList event now contains the following fields: WaitMarked, + EndMarked, and Waiting. This displays additional properties about the + user's profile, as well as whether or not the user is waiting for a + Marked user to enter the conference. + + * Added a new option for conference recording, record_file_append. If enabled, + when the recording is stopped and then re-started, the existing recording + will be used and appended to. + + * ConfBridge now has the ability to set the language of announcements to the + conference. The language can be set on a bridge profile in confbridge.conf + or by the dialplan function CONFBRIDGE(bridge,language)=en. + +ControlPlayback +------------------ + * The channel variable CPLAYBACKSTATUS may now return the value + 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface, + such as AMI. See the AMI action ControlPlayback for more information. + +Directory +------------------ + * Added the 'a' option, which allows the caller to enter in an additional + alias for the user in the directory. This option must be used in conjunction + with the 'f', 'l', or 'b' options. Note that the alias for a user can be + specified in voicemail.conf. + +DumpChan +------------------ + * The output of DumpChan no longer includes the DirectBridge or IndirectBridge + fields. Instead, if a channel is in a bridge, it includes a BridgeID field + containing the unique ID of the bridge that the channel happens to be in. + +ForkCDR +------------------ + * ForkCDR no longer automatically resets the forked CDR. See the 'r' option + for more information. + + * Variables are no longer purged from the original CDR. See the 'v' option for + more information. + + * The 'A' option has been removed. The Answer time on a CDR is never updated + once set. + + * The 'd' option has been removed. The disposition on a CDR is a function of + the state of the channel and cannot be altered. + + * The 'D' option has been removed. Who the Party B is on a CDR is a function + of the state of the respective channels involved in the CDR and cannot be + altered. + + * The 'r' option has been changed. Previously, ForkCDR always reset the CDR + such that the start time and, if applicable, the answer time was updated. + Now, by default, ForkCDR simply forks the CDR, maintaining any times. The + 'r' option now triggers the Reset, setting the start time (and answer time + if applicable) to the current time. Note that the 'a' option still sets + the answer time to the current time if the channel was already answered. + + * The 's' option has been removed. A variable can be set on the original CDR + if desired using the CDR function, and removed from a forked CDR using the + same function. + + * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no + longer applies in the CDR engine. + + * The 'v' option now prevents the copy of the variables from the original CDR + to the forked CDR. Previously the variables were always copied but were + removed from the original. This was changed as removing variables from a CDR + can have unintended side effects - this option allows the user to prevent + propagation of variables from the original to the forked without modifying + the original. + +MeetMe +------------------- + * Added the 'n' option to MeetMe to prevent application of the DENOISE + function to a channel joining a conference. Some channel drivers that vary + the number of audio samples in a voice frame will experience significant + quality problems if a denoiser is attached to the channel; this option gives + them the ability to remove the denoiser without having to unload func_speex. + +MixMonitor +------------------ + * The 'b' option now includes conferences as well as sounds played to the + participants. + + * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor + running during a transfer. If a MixMonitor is started on a channel, + the MixMonitor will continue to record the audio passing through the + channel even in the presence of transfers. + +NoCDR +------------------ + * The NoCDR application is deprecated. Please use the CDR_PROP function to + disable CDRs. + + * While the NoCDR application will prevent CDRs for a channel from being + propagated to registered CDR backends, it will not prevent that data from + being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP + function that enables CDRs on a channel will restore those records that have + not yet been finalized. + +ParkAndAnnounce +------------------- + * The app_parkandannounce module has been removed. The application + ParkAndAnnounce is now provided by the res_parking module. See the + res_parking changes for more information. + +Queue +------------------- + * Added queue available hint. The hint can be added to the dialplan using the + following syntax: exten,hint,Queue:{queue_name}_avail + For example, if the name of the queue is 'markq': + exten => 8501,hint,Queue:markq_avail + This will report 'InUse' if there are no logged in agents or no free agents. + It will report 'Idle' when an agent is free. + + * Queues now support a hint for member paused state. The hint uses the form + 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name} + are the name of the queue and the name of the member to subscribe to, + respectively. For example: exten => 8501,hint,Queue:sales_pause_mark. + Members will show as In Use when paused. + + * The configuration options eventwhencalled and eventmemberstatus have been + removed. As a result, the AMI events QueueMemberStatus, AgentCalled, + AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be + sent. The "Variable" fields will also no longer exist on the Agent* events. + These events can be filtered out from a connected AMI client using the + eventfilter setting in manager.conf. + + * The queue log now differentiates between blind and attended transfers. A + blind transfer will result in a BLINDTRANSFER message with the destination + context and extension. An attended transfer will result in an + ATTENDEDTRANSFER message. This message will indicate the method by which + the attended transfer was completed: "BRIDGE" for a bridge merge, "APP" + for running an application on a bridge or channel, or "LINK" for linking + two bridges together with local channels. The queue log will also now detect + externally initiated blind and attended transfers and record the transfer + status accordingly. + + * When performing queue pause/unpause on an interface without specifying an + individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at + least one member of any queue exists for that interface. + + * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT + for realtime queue log entries. + +ResetCDR +------------------ + * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable + CDRs when they were previously disabled on a channel. + + * The 'w' and 'a' options have been removed. Dispatching CDRs to registered + backends occurs on an as-needed basis in order to preserve linkedid + propagation and other needed behavior. + +SayAlphaCase +------------------ + * A new application, this is similar to SayAlpha except that it supports + case sensitive playback of the specified characters. For example, + SayAlphaCase(u,aBc) will result in 'a uppercase b c'. + +SetAMAFlags +------------------ + * This application is deprecated in favor of CHANNEL(amaflags). + +SendDTMF +------------------ + * The SendDTMF application will now accept 'W' as valid input. This will cause + the application to delay one second while streaming DTMF. + +Stasis +------------------ + * A new application in Asterisk 12, this hands control of the channel calling + the application over to an external system. Currently, external systems + manipulate channels in Stasis through the Asterisk RESTful Interface (ARI). + +UserEvent +------------------ + * UserEvent will now handle duplicate keys by overwriting the previous value + assigned to the key. + + * In addition to AMI, UserEvent invocations will now be distributed to any + interested Stasis applications. + +VoiceMail +------------------ + * Mailboxes defined by app_voicemail MUST be referenced by the rest of the + system as mailbox@context. The rest of the system cannot add @default + to mailbox identifiers for app_voicemail that do not specify a context + any longer. It is a mailbox identifier format that should only be + interpreted by app_voicemail. + + * The voicemail.conf configuration file now has an 'alias' configuration + parameter for use with the Directory application. The voicemail realtime + database table schema has also been updated with an 'alias' column. + + +Codecs +------------------ + * Pass through support has been added for both VP8 and Opus. + + * Added format attribute negotiation for the Opus codec. Format attribute + negotiation is provided by the res_format_attr_opus module. + + +Core +------------------ + * Masquerades as an operation inside Asterisk have been effectively hidden + by the migration to the Bridging API. As such, many 'quirks' of Asterisk + no longer occur. This includes renaming of channels, "" channels, + dropping of frame/audio hooks, and other internal implementation details + that users had to deal with. This fundamental change has large implications + throughout the changes documented for this version. For more information + about the new core architecture of Asterisk, please see the Asterisk wiki. + + * Multiple parties in a bridge may now be transferred. If a participant in a + multi-party bridge initiates a blind transfer, a Local channel will be used + to execute the dialplan location that the transferer sent the parties to. If + a participant in a multi-party bridge initiates an attended transfer, + several options are possible. If the attended transfer results in a transfer + to an application, a Local channel is used. If the attended transfer results + in a transfer to another channel, the resulting channels will be merged into + a single bridge. + + * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel + driver specific. If the channel variable is set on the transferrer channel, + the sound will be played to the target of an attended transfer. + + * The channel variable BRIDGEPEER becomes a comma separated list of peers in + a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers + listed. Any more peers in the bridge will not be included in the list. + BRIDGEPEER is not valid in holding bridges like parking since those channels + do not talk to each other even though they are in a bridge. + + * The channel variable BRIDGEPVTCALLID is only valid for two party bridges + and will contain a value if the BRIDGEPEER's channel driver supports it. + + * A channel variable ATTENDEDTRANSFER is now set which indicates which channel + was responsible for an attended transfer in a similar fashion to + BLINDTRANSFER. + + * Modules using the Configuration Framework or Sorcery must have XML + configuration documentation. This configuration documentation is included + with the rest of Asterisk's XML documentation, and is accessible via CLI + commands. See the CLI changes for more information. + +AMI (Asterisk Manager Interface) +------------------ + * Major changes were made to both the syntax as well as the semantics of the + AMI protocol. In particular, AMI events have been substantially improved + in this version of Asterisk. For more information, please see the AMI + specification at https://wiki.asterisk.org/wiki/x/dAFRAQ + + * AMI events that reference a particular channel or bridge will now always + contain a standard set of fields. When multiple channels or bridges are + referenced in an event, fields for at least some subset of the channels + and bridges in the event will be prefixed with a descriptive name to avoid + name collisions. See the AMI event documentation on the Asterisk wiki for + more information. + + * The CLI command 'manager show commands' no longer truncates command names + longer than 15 characters and no longer shows authorization requirement + for commands. 'manager show command' now displays the privileges needed + for using a given manager command instead. + + * The SIPshowpeer action will now include a 'SubscribeContext' field for a + peer in its response if the peer has a subscribe context set. + + * The SIPqualifypeer action now acknowledges the request once it has + established that the request is against a known peer. It also issues a new + event, 'SIPQualifyPeerDone', once the qualify action has been completed. + + * The PlayDTMF action now supports an optional 'Duration' parameter. This + specifies the duration of the digit to be played, in milliseconds. + + * Added VoicemailRefresh action to allow an external entity to trigger mailbox + updates when changes occur instead of requiring the use of pollmailboxes. + + * Added a new action 'ControlPlayback'. The ControlPlayback action allows an + AMI client to manipulate audio currently being played back on a channel. The + supported operations depend on the application being used to send audio to + the channel. When the audio playback was initiated using the ControlPlayback + application or CONTROL STREAM FILE AGI command, the audio can be paused, + stopped, restarted, reversed, or skipped forward. When initiated by other + mechanisms (such as the Playback application), the audio can be stopped, + reversed, or skipped forward. + + * Channel related events now contain a snapshot of channel state, adding new + fields to many of these events. + + * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed + in a future release. Please use the common 'Exten' field instead. + + * The AMI event 'UserEvent' from app_userevent now contains the channel state + fields. The channel state fields will come before the body fields. + + * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and + 'UnParkedCall' have changed significantly in the new res_parking module. + + The 'Channel' and 'From' headers are gone. For the channel that was parked + or is coming out of parking, a 'Parkee' channel snapshot is issued and it + has a number of fields associated with it. The old 'Channel' header relayed + the same data as the new 'ParkeeChannel' header. + + The 'From' field was ambiguous and changed meaning depending on the event. + for most of these, it was the name of the channel that parked the call + (the 'Parker'). There is no longer a header that provides this channel name, + however the 'ParkerDialString' will contain a dialstring to redial the + device that parked the call. + + On UnParkedCall events, the 'From' header would instead represent the + channel responsible for retrieving the parkee. It receives a channel + snapshot labeled 'Retriever'. The 'from' field is is replaced with + 'RetrieverChannel'. + + Lastly, the 'Exten' field has been replaced with 'ParkingSpace'. + + * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar + fashion has changed the field names 'StartExten' and 'StopExten' to + 'StartSpace' and 'StopSpace' respectively. + + * The deprecated use of | (pipe) as a separator in the channelvars setting in + manager.conf has been removed. + + * Channel Variables conveyed with a channel no longer contain the name of the + channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now + ChanVariable: bar=baz. When multiple channels are present in a single AMI + event, the various ChanVariable fields will contain a suffix that specifies + which channel they correspond to. + + * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI + event always conveys the AMI event for a particular channel. + + * All 'Reload' events have been consolidated into a single event type. This + event will always contain a Module field specifying the name of the module + and a Status field denoting the result of the reload. All modules now issue + this event when being reloaded. + + * The 'ModuleLoadReport' event has been removed. Most AMI connections would + fail to receive this event due to being connected after modules have loaded. + AMI connections that want to know when Asterisk is ready should listen for + the 'FullyBooted' event. + + * app_fax now sends the same send fax/receive fax events as res_fax. The + 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is + now the 'ReceiveFAX' event. + + * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and + 'MusicOnHoldStop'. The sub type field has been removed. + + * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a + carrier for another protocol. + + * The Bridge Manager action's 'Playtone' header now accepts more fine-grained + options. 'Channel1' and 'Channel2' may be specified in order to play a tone + to the specific channel. 'Both' may be specified to play a tone to both + channels. The old 'yes' option is still accepted as a way of playing the + tone to Channel2 only. + + * The AMI 'Status' response event to the AMI Status action replaces the + 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to + indicate what bridge the channel is currently in. + + * The AMI 'Hold' event has been moved out of individual channel drivers, into + core, and is now two events: 'Hold' and 'Unhold'. The status field has been + removed. + + * The AMI events in app_queue have been made more consistent with each other. + Events that reference channels (QueueCaller* and Agent*) will show + information about each channel. The (infamous) 'Join' and 'Leave' AMI + events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'. + + * The 'MCID' AMI event now publishes a channel snapshot when available and + its non-channel-snapshot parameters now use either the "MCallerID" or + 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead + of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named + parameters in the channel snapshot. + + * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed + 'AgentLogin' and 'AgentLogoff' respectively. + + * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been + renamed "DAHDIChannel" since it does not convey an Asterisk channel name. + + * 'ChannelUpdate' events have been removed. + + * All AMI events now contain a 'SystemName' field, if available. + + * Local channel optimization is now conveyed in two events: + 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent + when the Local channel driver begins attempting to optimize itself out of + the media path; the End event is sent after the channel halves have + successfully optimized themselves out of the media path. + + * Local channel information in events is now prefixed with 'LocalOne' and + 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of + the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin', + and 'LocalOptimizationEnd' events. + + * The option 'allowmultiplelogin' can now be set or overriden in a particular + account. When set in the general context, it will act as the default + setting for defined accounts. + + * The 'BridgeAction' event was removed. It technically added no value, as the + Bridge Action already receives confirmation of the bridge through a + successful completion Event. + + * The 'BridgeExec' events were removed. These events duplicated the events that + occur in the Bridging API, and are conveyed now through BridgeCreate, + BridgeEnter, and BridgeLeave events. + + * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from + previous versions. They now report all SR/RR packets sent/received, and + have been restructured to better reflect the data sent in a SR/RR. In + particular, the event structure now supports multiple report blocks. + + * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are + raised when a blind transfer/attended transfer completes successfully. + They contain information about the transfer that just completed, including + the location of the transfered channel. + + * Added a 'security' class to AMI which outputs the required fields for + security messages similar to the log messages from res_security_log + + * The AMI event 'ExtensionStatus' now contains a 'StatusText' field + that describes the status value in a human readable string. + +CDR (Call Detail Records) +------------------ + * Significant changes have been made to the behavior of CDRs. The CDR engine + was effectively rewritten and built on the Stasis message bus. For a full + definition of CDR behavior in Asterisk 12, please read the specification + on the Asterisk wiki (wiki.asterisk.org). + + * CDRs will now be created between all participants in a bridge. For each + pair of channels in a bridge, a CDR is created to represent the path of + communication between those two endpoints. This lets an end user choose who + to bill for what during bridge operations with multiple parties. + + * The duration, billsec, start, answer, and end times now reflect the times + associated with the current CDR for the channel, as opposed to a cumulative + measurement of all CDRs for that channel. + + * When a CDR is dispatched, user defined CDR variables from both parties are + included in the resulting CDR. If both parties have the same variable, only + the Party A value is provided. + + * Added a new option to cdr.conf, 'debug'. When enabled, significantly more + information regarding the CDR engine is logged as verbose messages. This + option should only be used if the behavior of the CDR engine needs to be + debugged. + + * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting + normally configured in cdr.conf. + + * Added CLI command 'cdr show active {channel}'. When {channel} is not + specified, this command provides a summary of the channels with CDR + information and their statistics. When {channel} is specified, it shows + detailed information about all records associated with {channel}. + +CEL (Channel Event Logging) +------------------ + * CEL has undergone significant rework in Asterisk 12, and is now built on the + Stasis message bus. Please see the specification for CEL on the Asterisk + wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed + information. + + * The 'extra' field of all CEL events that use it now consists of a JSON blob + with key/value pairs which are defined in the Asterisk 12 CEL documentation. + + * BLINDTRANSFER events now report the transferee bridge unique + identifier, extension, and context in a JSON blob as the extra string + instead of the transferee channel name as the peer. + + * ATTENDEDTRANSFER events now report the peer as NULL and additional + information in the 'extra' string as a JSON blob. For transfers that occur + between two bridged channels, the 'extra' JSON blob contains the primary + bridge unique identifier, the secondary channel name, and the secondary + bridge unique identifier. For transfers that occur between a bridged channel + and a channel running an app, the 'extra' JSON blob contains the primary + bridge unique identifier, the secondary channel name, and the app name. + + * LOCAL_OPTIMIZE events have been added to convey local channel + optimizations with the record occurring for the semi-one channel and + the semi-two channel name in the peer field. + + * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER, + CONF_EXIT, CONF_START, and CONF_END events have all been removed. These + events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER + and BRIDGE_EXIT events are raised when a channel enters/exits any bridge, + regardless of whether or not that bridge happens to contain multiple + parties. + +CLI +------------------- + * When compiled with '--enable-dev-mode', the astobj2 library will now add + several CLI commands that allow for inspection of ao2 containers that + register themselves with astobj2. The CLI commands are 'astobj2 container + dump', 'astobj2 container stats', and 'astobj2 container check'. + + * Added specific CLI commands for bridge inspection. This includes 'bridge + show all', which lists all bridges in the system, and 'bridge show {id}', + which provides specific information about a bridge. + + * Added CLI command 'bridge destroy'. This will destroy the specified bridge, + ejecting the channels currently in the bridge. If the channels cannot + continue in the dialplan or application that put them in the bridge, they + will be hung up. + + * Added command 'bridge kick'. This will eject a single channel from a bridge. + + * Added commands to inspect and manipulate the registered bridge technologies. + This include 'bridge technology show', which lists the registered bridge + technologies, as well as 'bridge technology {suspend|unsuspend} {tech}', + which controls whether or not a registered bridge technology can be used + during smart bridge operations. If a technology is suspended, it will not + be used when a bridge technology is picked for channels; when unsuspended, + it can be used again. + + * The command 'config show help {module} {type} {option}' will show + configuration documentation for modules with XML configuration + documentation. When {module}, {type}, and {option} are omitted, a listing + of all modules with registered documentation is displayed. When {module} + is specified, a listing of all configuration types for that module is + displayed, along with their synopsis. When {module} and {type} are + specified, a listing of all configuration options for that type are + displayed along with their synopsis. When {module}, {type}, and {option} + are specified, detailed information for that configuration option is + displayed. + + * Added 'core show sounds' and 'core show sound' CLI commands. These display + a listing of all installed media sounds available on the system and + detailed information about a sound, respectively. + + * 'xmldoc dump' has been added. This CLI command will dump the XML + documentation DOM as a string to the specified file. The Asterisk core + will populate certain XML elements pulled from the source files with + additional run-time information; this command lets a user produce the + XML documentation with all information. + +Features +------------------- + * Parking has been pulled from core and placed into a separate module called + res_parking. See Parking changes below for more details. Configuration for + parking should now be performed in res_parking.conf. Configuration for + parking in features.conf is now unsupported. + + * Core attended transfers now have several new options. While performing an + attended transfer, the transferer now has the following options: + - *1 - cancel the attended transfer (configurable via atxferabort) + - *2 - complete the attended transfer, dropping out of the call + (configurable via atxfercomplete) + - *3 - complete the attended transfer, but stay in the call. This will turn + the call into a multi-party bridge (configurable via atxferthreeway) + - *4 - swap to the other party. Once an attended transfer has begun, this + options may be used multiple times (configurable via atxferswap) + + * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT + must be on the channel initiating the transfer to have any effect. + + * The BRIDGE_FEATURES channel variable would previously only set features for + the calling party and would set this feature regardless of whether the + feature was in caps or in lowercase. Use of a caps feature for a letter + will now apply the feature to the calling party while use of a lowercase + letter will apply that feature to the called party. + + * Add support for automixmon to the BRIDGE_FEATURES channel variable. + + * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is + removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that + activated the dynamic feature. + + * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set + only on the channel executing the dynamic feature. Executing a dynamic + feature on the bridge peer in a multi-party bridge will execute it on all + peers of the activating channel. + + * You can now have the settings for a channel updated using the FEATURE() + and FEATUREMAP() functions inherited to child channels by setting + FEATURE(inherit)=yes. + + * automixmon now supports additional channel variables from automon including: + TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START, + and TOUCH_MIXMONITOR_MESSAGE_STOP + + * A new general features.conf option 'recordingfailsound' has been added which + allowssetting a failure sound for a user tries to invoke a recording feature + such as automon or automixmon and it fails. + + * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in + features.c for atxferdropcall=no to work properly. This option now just + works. + +Logging +------------------- + * Added log rotation strategy 'none'. If set, no log rotation strategy will + be used. Given that this can cause the Asterisk log files to grow quickly, + this option should only be used if an external mechanism for log management + is preferred. + +Realtime +------------------ + * Dynamic realtime tables for SIP Users can now include a 'path' field. This + will store the path information for that peer when it registers. Realtime + tables can also use the 'supportpath' field to enable Path header support. + + * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport + objectIdentifier. This maps to the supportpath option in sip.conf. + +Sorcery +------------------ + * Sorcery is a new data abstraction and object persistence API in Asterisk. It + provides modules a useful abstraction on top of the many storage mechanisms + in Asterisk, including the Asterisk Database, static configuration files, + static Realtime, and dynamic Realtime. It also provides a caching service. + Users can configure a hierarchy of data storage layers for specific modules + in sorcery.conf. + + * All future modules which utilize Sorcery for object persistence must have a + column named "id" within their schema when using the Sorcery realtime module. + This column must be able to contain a string of up to 128 characters in length. + +Security Events Framework +------------------ + * Security Event timestamps now use ISO 8601 formatted date/time instead of + the "seconds-microseconds" format that it was using previously. + +Stasis Message Bus +------------------ + * The Stasis message bus is a publish/subscribe message bus internal to + Asterisk. Many services in Asterisk are built on the Stasis message bus, + including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of + Stasis can be configured in stasis.conf. Note that these parameters operate + at a very low level in Asterisk, and generally will not require changes. + +Channel Drivers +------------------ + * When a channel driver is configured to enable jiterbuffers, they are now + applied unconditionally when a channel joins a bridge. If a jitterbuffer + is already set for that channel when it enters, such as by the JITTERBUFFER + function, then the existing jitterbuffer will be used and the one set by + the channel driver will not be applied. + +chan_agent +------------------ + * chan_agent has been removed and replaced with AgentLogin and AgentRequest + dialplan applications provided by the app_agent_pool module. Agents are + connected with callers using the new AgentRequest dialplan application. + The Agents: device state is available to monitor the status of an + agent. See agents.conf.sample for valid configuration options. + + * The updatecdr option has been removed. Altering the names of channels on a + CDR is not supported - the name of the channel is the name of the channel, + and pretending otherwise helps no one. The AGENTUPDATECDR channel variable + has also been removed, for the same reason. + + * The endcall and enddtmf configuration options are removed. Use the + dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent + channel before calling AgentLogin. + +chan_bridge +------------------ + * chan_bridge has been removed. Its functionality has been incorporated + directly into the ConfBridge application itself. + +chan_dahdi +------------------ + * Added the CLI command 'pri destroy span'. This will destroy the D-channel + of the specified span and its B-channels. Note that this command should + only be used if you understand the risks it entails. + + * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'. + A range of channels can be specified to be destroyed. Note that this command + should only be used if you understand the risks it entails. + + * Added the CLI command 'dahdi create channels'. A range of channels can be + specified to be created, or the keyword 'new' can be used to add channels + not yet created. + + * The script specified by the chan_dahdi.conf mwimonitornotify option now gets + the exact configured mailbox name. For app_voicemail mailboxes this is + mailbox@context. + + * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled. + +chan_iax2 +------------------ + * IPv6 support has been added. We are now able to bind to and + communicate using IPv6 addresses. + +chan_local +------------------ + * The /b option has been removed. + + * chan_local moved into the system core and is no longer a loadable module. + +chan_mobile +------------------ + * Added general support for busy detection. + + * Added ECAM command support for Sony Ericsson phones. + +chan_pjsip +------------------ + * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP + SIP stack. A collection of resource modules provides the bulk of the SIP + functionality. For more information on the new SIP channel driver, see + https://wiki.asterisk.org/wiki/x/JYGLAQ + +chan_sip +------------------ + * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf + using the 'supportpath' setting, either on a global basis or on a peer basis. + This setting enables Asterisk to route outgoing out-of-dialog requests via a + set of proxies by using a pre-loaded route-set defined by the Path headers in + the REGISTER request. See Realtime updates for more configuration information. + + * The SIP_CODEC family of variables may now specify more than one codec. Each + codec must be separated by a comma. The first codec specified is the + preferred codec for the offer. This allows a dialplan writer to specify both + audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264) + + * The 'callevents' parameter has been removed. Hold AMI events are now raised + in the core, and can be filtered out using the 'eventfilter' parameter + in manager.conf. + + * Added 'ignore_requested_pref'. When enabled, this will use the preferred + codecs configured for a peer instead of the requested codec. + + * The option "register_retry_403" has been added to chan_sip to work around + servers that are known to erroneously send 403 in response to valid + REGISTER requests and allows Asterisk to continue attepmting to connect. + +chan_skinny +------------------ + * Added the 'immeddialkey' parameter. If set, when the user presses the + configured key the already entered number will be immediately dialed. This + is useful when the dialplan allows for variable length pattern matching. + Valid options are '*' and '#'. + + * Added the 'callfwdtimeout' parameter. This configures the amount of time (in + milliseconds) before a call forward is considered to not be answered. + + * The 'serviceurl' parameter allows Service URLs to be attached to line + buttons. + + +Functions +------------------ + +AGENT +------------------ + * The password option has been disabled, as the AgentLogin application no + longer provides authentication. + +AUDIOHOOK_INHERIT +------------------ + * Due to changes in the Asterisk core, this function is no longer needed to + preserve a MixMonitor on a channel during transfer operations and dialplan + execution. It is effectively obsolete. + +CDR (function) +------------------ + * The 'amaflags' and 'accountcode' attributes for the CDR function are + deprecated. Use the CHANNEL function instead to access these attributes. + + * The 'l' option has been removed. When reading a CDR attribute, the most + recent record is always used. When writing a CDR attribute, all non-finalized + CDRs are updated. + + * The 'r' option has been removed, for the same reason as the 'l' option. + + * The 's' option has been removed, as LOCKED semantics no longer exist in the + CDR engine. + +CDR_PROP +------------------ + * A new function CDR_PROP has been added. This function lets you set properties + on a channel's active CDRs. This function is write-only. Properties accept + boolean values to set/clear them on the channel's CDRs. Valid properties + include: + - 'party_a' - make this channel the preferred Party A in any CDR between two + channels. If two channels have this property set, the creation time of the + channel is used to determine who is Party A. Note that dialed channels are + never Party A in a CDR. + - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR + application when set to True, and analogous to the 'e' option in ResetCDR + when set to False. + +CHANNEL +------------------ + * Added the argument 'dtmf_features'. This sets the DTMF features that will be + enabled on a channel when it enters a bridge. Allowed values are 'T', 'K', + 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial + application. + + * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto + string, i.e., [[context],extension],priority. If set on a channel, if a + channel leaves a bridge but is not hung up it will resume dialplan execution + at that location. + +JITTERBUFFER +------------------ + * JITTERBUFFER now accepts an argument of 'disabled' which can be used + to remove jitterbuffers previously set on a channel with JITTERBUFFER. + The value of this setting is ignored when disabled is used for the argument. + +PJSIP_DIAL_CONTACTS +------------------ + * A new function provided by chan_pjsip, this function can be used in + conjunction with the Dial application to construct a dial string that will + dial all contacts on an Address of Record associated with a chan_pjsip + endpoint. + +PJSIP_MEDIA_OFFER +------------------ + * Provided by chan_pjsip, this function sets the codecs to be offered on the + outbound channel prior to dialing. + +REDIRECTING +------------------ + * Redirecting reasons can now be set to arbitrary strings. This means + that the REDIRECTING dialplan function can be used to set the redirecting + reason to any string. It also allows for custom strings to be read as the + redirecting reason from SIP Diversion headers. + +SPEECH_ENGINE +------------------ + * The SPEECH_ENGINE function now supports read operations. When read from, it + will return the current value of the requested attribute. + +VMCOUNT: +------------------ + * Mailboxes defined by app_voicemail MUST be referenced by the rest of the + system as mailbox@context. The rest of the system cannot add @default + to mailbox identifiers for app_voicemail that do not specify a context + any longer. It is a mailbox identifier format that should only be + interpreted by app_voicemail. + + +Resources +------------------ + +res_agi (Asterisk Gateway Interface) +------------------ + * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd. + + * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec, + and AsyncAGIEnd. + + * The CONTROL STREAM FILE command now accepts an offsetms parameter. This + will start the playback of the audio at the position specified. It will + also return the final position of the file in 'endpos'. + + * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS + channel variable if the user stopped the file playback or if a remote + entity stopped the playback. If neither stopped the playback, it will + indicate the overall success/failure of the playback. If stopped early, + the final offset of the file will be set in the CPLAYBACKOFFSET channel + variable. + + * The SAY ALPHA command now accepts an additional parameter to control + whether it specifies the case of uppercase, lowercase, or all letters to + provide functionality similar to SayAlphaCase. + +res_ari (Asterisk RESTful Interface) (and others) +------------------ + * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and + control telephony primitives in Asterisk by remote client. This includes + channels, bridges, endpoints, media, and other fundamental concepts. Users + of ARI can develop their own communications applications, controlling + multiple channels using an HTTP RESTful interface and receiving JSON events + about the objects via a WebSocket connection. ARI can be configured in + Asterisk via ari.conf. For more information on ARI, see + https://wiki.asterisk.org/wiki/x/0YCLAQ + +res_parking +------------------- + * Parking has been extracted from the Asterisk core as a loadable module, + res_parking. Configuration for parking is now provided by res_parking.conf. + Configuration through features.conf is no longer supported. + + * res_parking uses the configuration framework. If an invalid configuration is + supplied, res_parking will fail to load or fail to reload. Previously, + invalid configurations would generally be accepted, with certain errors + resulting in individually disabled parking lots. + + * Parked calls are now placed in bridges. While this is largely an + architectural change, it does have implications on how channels in a parking + lot are viewed. For example, commands that display channels in bridges will + now also display the channels in a parking lot. + + * The order of arguments for the new parking applications have been modified. + Timeout and return context/exten/priority are now implemented as options, + while the name of the parking lot is now the first parameter. See the + application documentation for Park, ParkedCall, and ParkAndAnnounce for more + in-depth information as well as syntax. + + * Extensions are by default no longer automatically created in the dialplan to + park calls or pickup parked calls. Generation of dialplan extensions can be + enabled using the 'parkext' configuration option. + + * ADSI functionality for parking is no longer supported. The 'adsipark' + configuration option has been removed as a result. + + * The PARKINGSLOT channel variable has been deprecated in favor of + PARKING_SPACE to match the naming scheme of the new system. + + * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked + channel even when the configuration option 'comebactoorigin' is enabled. + + * A new CLI command 'parking show' has been added. This allows a user to + inspect the parking lots that are currently in use. + 'parking show ' will also show the parked calls in a specific + parking lot. + + * The CLI command 'parkedcalls' is now deprecated in favor of + 'parking show '. + + * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which + can be used to get a list of parked calls for a specific parking lot. + + * The AMI command 'Park' field 'Channel2' has been deprecated and replaced + with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are + specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no + longer a required argument. + + * The ParkAndAnnounce application is now provided through res_parking instead + of through the separate app_parkandannounce module. + + * ParkAndAnnounce will no longer go to the next position in dialplan on timeout + by default. Instead, it will follow the timeout rules of the parking lot. The + old behavior can be reproduced by using the 'c' option. + + * Dynamic parking lots will now fail to be created under the following + conditions: + - if the parking lot specified by PARKINGDYNAMIC does not exist + - if they require exclusive park and parkedcall extensions which overlap + with existing parking lots. + + * Dynamic parking lots will be cleared on reload for dynamic parking lots that + currently contain no calls. Dynamic parking lots containing parked calls + will persist through the reloads without alteration. + + * If 'parkext_exclusive' is set for a parking lot and that extension is + already in use when that parking lot tries to register it, this is now + considered a parking system configuration error. Configurations which do + this will be rejected. + + * Added channel variable PARKER_FLAT. This contains the name of the extension + that would be used if 'comebacktoorigin' is enabled. This can be useful when + comebacktoorigin is disabled, but the dialplan or an external control + mechanism wants to use the extension in the park-dial context that was + generated to re-dial the parker on timeout. + +res_pjsip (and many others) +------------------ + * A large number of resource modules make up the SIP stack based on pjsip. + The chan_pjsip channel driver users these resource modules to provide + various SIP functionality in Asterisk. The majority of configuration for + these modules is performed in pjsip.conf. Other modules may use their + own configuration files. + + * Added 'set_var' option for an endpoint. For each variable specified that + variable gets set upon creation of a channel involving the endpoint. + +res_rtp_asterisk +------------------ + * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable + them, an Asterisk-specific version of PJSIP needs to be installed. + Tarballs are available from https://github.com/asterisk/pjproject/tags/. + +res_statsd/res_chan_stats +------------------ + * A new resource module, res_statsd, has been added, which acts as a statsd + client. This module allows Asterisk to publish statistics to a statsd + server. In conjunction with res_chan_stats, it will publish statistics about + channels to the statsd server. It can be configured via res_statsd.conf. + +res_xmpp +------------------ + * Device state for XMPP buddies is now available using the following format: + XMPP// + If any resource is available the device state is considered to be not in use. + If no resources exist or all are unavailable the device state is considered + to be unavailable. + + +Scripts +------------------ + +Realtime/Database Scripts +------------------ + * Asterisk previously included example db schemas in the contrib/realtime/ + directory of the source tree. This has been replaced by a set of database + migrations using the Alembic framework. This allows you to use alembic to + initialize the database for you. It will also serve as a database migration + tool when upgrading Asterisk in the future. + + See contrib/ast-db-manage/README.md for more details. + +sip_to_res_pjsip.py +------------------- + * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder. + This python script will convert an existing sip.conf file to a + pjsip.conf file, for use with the chan_pjsip channel driver. This script + is meant to be an aid in converting an existing chan_sip configuration to + a chan_pjsip configuration, but it is expected that configuration beyond + what the script provides will be needed. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 10 to Asterisk 11 -------------------- +------------------------------------------------------------------------------ + +Build System +------------------- + * The Asterisk build system will now build and install a shared library + (libasteriskssl.so) used to wrap various initialization and shutdown functions + from the libssl and libcrypto libraries provided by OpenSSL. This is done so + that Asterisk can ensure that these functions do *not* get called by any + modules that are loaded into Asterisk, since they should only be called once + in any single process. If desired, this feature can be disabled by supplying + the "--disable-asteriskssl" option to the configure script. + + * A new make target, 'full', has been added to the Makefile. This performs + the same compilation actions as make all, but will also scan the entirety of + each source file for documentation. This option is needed to generate AMI + event documentation. Note that your system must have Python in order for + this make target to succeed. + + * The optimization portion of the build system has been reworked to avoid + broken builds on certain architectures. All architecture-specific + optimization has been removed in favor of using -march=native to allow gcc + to detect the environment in which it is running when possible. This can + be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect. + + * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g., + make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever" + + * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you + previously parsed the header file to obtain the version of Asterisk, you + will now have to go through Asterisk to get the version information. + + +Applications +------------------- + +Bridge +------------------- + * Added 'F()' option. Similar to the dial option, this can be supplied with + arguments indicating where the callee should go after the caller is hung up, + or without options specified, the priority after the Queue will be used. + + +ConfBridge +------------------- + * Added menu action admin_toggle_mute_participants. This will mute / unmute + all non-admin participants on a conference. The confbridge configuration + file also allows for the default sounds played to all conference users when + this occurs to be overriden using sound_participants_unmuted and + sound_participants_muted. + + * Added menu action participant_count. This will playback the number of + current participants in a conference. + + * Added announcement configuration option to user profile. If set the sound + file will be played to the user, and only the user, upon joining the + conference bridge. + + * Added record_file_append option that defaults to "yes", but if set to no + will create a new file between each start/stop recording. + + +Dial +------------------- + * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller + channels respectively before the callee channels are called. + + +ExternalIVR +------------------- + * Added support for IPv6. + + * Add interrupt ('I') command to ExternalIVR. Sending this command from an + external process will cause the current playlist to be cleared, including + stopping any audio file that is currently playing. This is useful when you + want to interrupt audio playback only when specific DTMF is entered by the + caller. + + +FollowMe +------------------- + * A new option, 'I' has been added to app_followme. By setting this option, + Asterisk will not update the caller with connected line changes when they + occur. This is similar to app_dial and app_queue. + + * The 'N' option is now ignored if the call is already answered. + + * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee + and caller channels respectively before the callee channels are called. + + * The winning FollowMe outgoing call is now put on hold if the caller put it on + hold. + + +MixMonitor +------------------ + * MixMonitor hooks now have IDs associated with them which can be used to + assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option + will allow storage of the MixMonitor ID in a channel variable. StopMixmonitor + now accepts that ID as an argument. + + * Added 'm' option, which stores a copy of the recording as a voicemail in the + indicated mailboxes. + + +MySQL +------------------- + * The connect action in app_mysql now allows you to specify a port number to + connect to. This is useful if you run a MySQL server on a non-standard + port number. + + +OSP Applications +------------------- + * Increased the default number of allowed destinations from 5 to 12. + + +Page +------------------- + * The app_page application now no longer depends on DAHDI or app_meetme. It + has been re-architected to use app_confbridge internally. + + +Queue +------------------- + * Added queue options autopausebusy and autopauseunavail for automatically + pausing a queue member when their device reports busy or congestion. + + * The 'ignorebusy' option for queue members has been deprecated in favor of + the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been + added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a + per interface basis. Individual ringinuse values can now be set in + queues.conf via an argument to member definitions. Lastly, the queue + 'ringinuse' setting now only determines defaults for the per member + 'ringinuse' setting and does not override per member settings like it does + in earlier versions. + + * Added 'F()' option. Similar to the dial option, this can be supplied with + arguments indicating where the callee should go after the caller is hung up, + or without options specified, the priority after the Queue will be used. + + * Added new option log_member_name_as_agent, which will cause the membername to + be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a + state_interface has been set. + + * Add queue monitoring hints. exten => 8501,hint,Queue:markq. + + * App_queue will now play periodic announcements for the caller that + holds the first position in the queue while waiting for answer. + +SayUnixTime +------------------ + * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension + when receiving DTMF. Use the 'j' option to enable extension jumping. Also + changed arguments to SayUnixTime so that every option is truly optional even + when using multiple options (so that j option could be used without having to + manually specify timezone and format) There are other benefits, e.g., format + can now be used without specifying time zone as well. + + +Voicemail +------------------ + * Addition of the VM_INFO function - see Function changes. + + * The imapserver, imapport, and imapflags configuration options can now be + overriden on a user by user basis. + + * When voicemail plays a message's envelope with saycid set to yes, when + reaching the caller id field it will play a recording of a file with the same + base name as the sender's callerid if there is a similarly named file in + /recordings/callerids/ + + * Voicemails now contains a unique message identifier "msg_id", which is stored + in the message envelope with the sound files. IMAP backends will now store + the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC + backends will store the message identifier in a "msg_id" column. See + UPGRADE.txt for more information. + + * Added VoiceMailPlayMsg application. This application will play a single + voicemail message from a mailbox. The result of the application, SUCCESS or + FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS. + + +Functions +------------------ + * Hangup handlers can be attached to channels using the CHANNEL() function. + Hangup handlers will run when the channel is hung up similar to the h + extension. The hangup_handler_push option will push a GoSub compatible + location in the dialplan onto the channel's hangup handler stack. The + hangup_handler_pop option will remove the last added location, and optionally + replace it with a new GoSub compatible location. The hangup_handler_wipe + option will remove all locations on the stack, and optionally add a new + location. + + * The expression parser now recognizes the ABS() absolute value function, + which will convert negative floating point values to positive values. + + * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan + control of faxdetect. + + * Addition of the VM_INFO function that can be used to retrieve voicemail + user information, such as the email address and full name. + The MAILBOX_EXISTS dialplan function has been deprecated in favour of + VM_INFO. + + * The REDIRECTING function now supports the redirecting original party id + and reason. + + * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() + lets you set some of the configuration options from the [general] section + of features.conf on a per-channel basis. FEATUREMAP() lets you customize + the key sequence used to activate built-in features, such as blindxfer, + and automon. See the built-in documentation for details. + + * MESSAGE(from) for incoming SIP messages now returns "display-name" + instead of simply the uri. This is the format that MessageSend() can use + in the from parameter for outgoing SIP messages. + + * Added the PRESENCE_STATE function. This allows retrieving presence state + information from any presence state provider. It also allows setting + presence state information from a CustomPresence presence state provider. + See AMI/CLI changes for related commands. + + * Added the AMI_CLIENT function to make manager account attributes available + to the dialplan. It currently supports returning the current number of + active sessions for a given account. + + * Added support for private party ID information to CALLERID, CONNECTEDLINE, + and the REDIRECTING functions. + + +Channel Drivers +------------------ + +chan_local +------------------ + * Added a manager event "LocalBridge" for local channel call bridges between + the two pseudo-channels created. + + +chan_dahdi +------------------ + * Added dialtone_detect option for analog ports to disconnect incoming + calls when dialtone is detected. + + * Added option colp_send to send ISDN connected line information. Allowed + settings are block, to not send any connected line information; connect, to + send connected line information on initial connect; and update, to send + information on any update during a call. Default is update. + + * Add options namedcallgroup and namedpickupgroup to support installations + where a higher number of groups (>64) is required. + + * Added support to use private party ID information with PRI calls. + + +chan_motif +------------------ + * A new channel driver named chan_motif has been added which provides support for + Google Talk and Jingle in a single channel driver. This new channel driver includes + support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, + hold, unhold, and ringing notification. It is also compliant with the current Jingle + specification, current Google Jingle specification, and the original Google Talk + protocol. + + +chan_ooh323 +------------------ + * Added NAT support for RTP. Setting in config is 'nat', which can be set + globally and overriden on a peer by peer basis. + + * Direct media functionality has been added. Options in config are: + directmedia (directrtp) and directrtpsetup (earlydirect) + + * ChannelUpdate events now contain a CallRef header. + + +chan_sip +------------------ + * Asterisk will no longer substitute CID number for CID name in the display + name field if CID number exists without a CID name. This change improves + compatibility with certain device features such as Avaya IP500's directory + lookup service. + + * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers + created using that setting to not be removed during SIP reload. + + * Added settings recordonfeature and recordofffeature. When receiving an INFO + request with a "Record:" header, this will turn the requested feature on/off. + Allowed values are 'automon', 'automixmon', and blank to disable. Note that + dynamic features must be enabled and configured properly on the requesting + channel for this to function properly. + + * Add support to realtime for the 'callbackextension' option. + + * When multiple peers exist with the same address, but differing + callbackextension options, incoming requests that are matched by address + will be matched to the peer with the matching callbackextension if it is + available. + + * Two new NAT options, auto_force_rport and auto_comedia, have been added + which set the force_rport and comedia options automatically if Asterisk + detects that an incoming SIP request crossed a NAT after being sent by + the remote endpoint. + + * The default global nat setting in sip.conf has been changed from force_rport + to auto_force_rport. + + * NAT settings are now a combinable list of options. The equivalent of the + deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before. + + * Adds an option send_diversion which can be disabled to prevent + diversion headers from automatically being added to INVITE requests. + + * Add support for lightweight NAT keepalive. If enabled a blank packet will + be sent to the remote host at a given interval to keep the NAT mapping open. + This can be enabled using the keepalive configuration option. + + * Add option 'tonezone' to specify country code for indications. This option + can be set both globally and overridden for specific peers. + + * The SIP Security Events Framework now supports IPv6. + + * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares + between multiple user agents. When set, for directmedia reinvites, + Asterisk will not send an immediate reinvite on an incoming call leg. This + option is useful when peered with another SIP user agent that is known to + send immediate direct media reinvites upon call establishment. + + * Add support for WebSocket transport. This can be configured using 'ws' or 'wss' + as the transport. + + * Add options subminexpiry and submaxexpiry to set limits of subscription + timer independently from registration timer settings. The setting of the + registration timer limits still is done by options minexpiry, maxexpiry + and defaultexpiry. For backwards compatibility the setting of minexpiry + and maxexpiry also is used to configure the subscription timer limits if + subminexpiry and submaxexpiry are not set in sip.conf. + + * Set registration timer limits to default values when reloading sip + configuration and values are not set by configuration. + + * Add options namedcallgroup and namedpickupgroup to support installations + where a higher number of groups (>64) is required. + + * When a MESSAGE request is received, the address the request was received from + is now saved in the SIP_RECVADDR variable. + + * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now + parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present, + the ANI2/OLI information is set on the channel, which can be retrieved using + the CALLERID function. + + * Peers can now be configured to support negotiation of ICE candidates using + the setting icesupport. See res_rtp_asterisk changes for more information. + + * Added support for format attribute negotiation. See the Codecs changes for + more information. + + * Extra headers specified with SIPAddHeader are sent with the REFER message + when using Transfer application. See refer_addheaders in sip.conf.sample. + + * Added support to use private party ID information with calls. + + * Adds an option discard_remote_hold_retrieval that when set stops telling + the peer to start music on hold. + + +chan_skinny +------------------ + * Added skinny version 17 protocol support. + + +chan_unistim +-------------------- + * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set + + * Modified option 'date_format' to allow options to display date in 31Jan and Jan31 + formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3 + as per the UNISTIM protocol. + + * Fixed issues with dialtone not matching indications.conf and mute stopping rx + as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e" + + * Added ability to use multiple lines for a single phone. This allows multiple + calls to occur on a single phone, using callwaiting and switching between calls. + + * Added option 'sharpdial' allowing end dialing by pressing # key + + * Added option 'interdigit_timer' to control phone dial timeout + + * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance + + * Added global 'debug' option, that enables debug in channel driver + + * Added ability to translate on-screen menu in multiple languages. Tested on + Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, + ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen + menu of phone + + * In addition to English added French and Russian languages for on-screen menus + + * Reworked dialing number input: added dialing by timeout, immediate dial on + on dialplan compare, phone number length now not limited by screen size + + * Added ability to pickup a call using features.conf defined value and + on-screen key + + +chan_mISDN: +------------------ + * Add options namedcallgroup and namedpickupgroup to support installations + where a higher number of groups (>64) is required. + + * Added support to use private party ID information with calls. + + +Core +------------------ + * The minimum DTMF duration can now be configured in asterisk.conf + as "mindtmfduration". The default value is (as before) set to 80 ms. + (previously it was only available in source code) + + * Named ACLs can now be specified in acl.conf and used in configurations that + use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is + used to specify an ACL, a similar form of 'acl' will add a named ACL to the + working ACL. In addition, some CLI commands have been added to provide + show information and allow for module reloading - see CLI Changes. + + * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple + items (separated by commas), and items in the rule can be negated by prefixing + them with '!'. This simplifies Asterisk Realtime configurations, since it is no + longer necessray to control the order that the 'permit' and 'deny' columns are + returned from queries. + + * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to + be used within the dynamic weight attribute when specifying a mapping. + + * CEL backends can now be configured to show "USER_DEFINED" in the EventName + header, instead of putting the user defined event name there. When enabled + the UserDefType header is added for user defined events. This feature is + enabled with the setting show_user_defined. + + * Macro has been deprecated in favor of GoSub. For redirecting and connected + line purposes use the following variables instead of their macro equivalents: + REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB, + CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of + cc_callback_macro in channel configurations. + + * Asterisk can now use a system-provided NetBSD editline library (libedit) if it + is available. + + * Call files now support the "early_media" option to connect with an outgoing + extension when early media is received. + + * Added support to use private party ID information with calls. + + +AGI +------------------ + * A new channel variable, AGIEXITONHANGUP, has been added which allows + Asterisk to behave like it did in Asterisk 1.4 and earlier where the + AGI application would exit immediately after a channel hangup is detected. + + * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames + are resolved and each address is attempted in turn until one succeeds or + all fail. + + +AMI (Asterisk Manager Interface) +------------------ + * The originate action now has an option "EarlyMedia" that enables the + call to bridge when we get early media in the call. Previously, + early media was disregarded always when originating calls using AMI. + + * Added setvar= option to manager accounts (much like sip.conf) + + * Originate now generates an error response if the extension given is not found + in the dialplan + + * MixMonitor will now show IDs associated with the mixmonitor upon creating + them if the i(variable) option is used. StopMixMonitor will accept + MixMonitorID as an option to close specific MixMonitors. + + * The SIPshowpeer manager action response field "SIP-Forcerport" has been + updated to include information about peers configured with + nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is + detected, and "a" if it is set and nat is not detected. "Y" and "N" are still + returned if auto_force_rport is not enabled. + + * Added SIPpeerstatus manager command which will generate PeerStatus events + similar to the existing PeerStatus events found in chan_sip on demand. + + * Hangup now can take a regular expression as the Channel option. If you want + to hangup multiple channels, use /regex/ as the Channel option. Existing + behavior to hanging up a single channel is unchanged, but if you pass a regex, + the manager will send you a list of channels back that were hung up. + + * Support for IPv6 addresses has been added. + + * AMI Events can now be documented in the Asterisk source. Note that AMI event + documentation is only generated when Asterisk is compiled using 'make full'. + See the CLI section for commands to display AMI event information. + + * The AMI Hangup event now includes the AccountCode header so you can easily + correlate with AMI Newchannel events. + + * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include + the StateInterface of the queue member. + + * Added AMI event SessionTimeout in the Call category that is issued when a + call is terminated due to either RTP stream inactivity or SIP session timer + expiration. + + * CEL events can now contain a user defined header UserDefType. See core + changes for more information. + + * OOH323 ChannelUpdate events now contain a CallRef header. + + * Added PresenceState command. This command will report the presence state for + the given presence provider. + + * Added Parkinglots command. This will list all parking lots as a series of + AMI Parkinglot events. + + * Added MessageSend command. This behaves in the same manner as the + MessageSend application, and is a technolgoy agnostic mechanism to send out + of call text messages. + + * Added "message" class authorization. This grants an account permission to + send out of call messages. Write-only. + + +CLI +------------------- + * The "dialplan add include" command has been modified to create context a context + if one does not already exist. For instance, "dialplan add include foo into bar" + will create context "bar" if it does not already exist. + + * A "dialplan remove context" command has been added to remove a context from + the dialplan + + * The "mixmonitor list " command will now show MixMonitor ID, and the + filenames of all running mixmonitors on a channel. + + * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if + numeric instead of 0, 1, or 2. + + * "stun show status" will show a table describing how the STUN client is + behaving. + + * "acl show [named acl]" will show information regarding a Named ACL. The + acl module can be reloaded with "reload acl". + + * Added CLI command to display AMI event information - "manager show events", + which shows a list of all known and documented AMI events, and "manager show + event [event name]", which shows detail information about a specific AMI + event. + + * The result of the CLI command "queue show" now includes the state interface + information of the queue member. + + * The command "core set verbose" will now set a separate level of logging for + each remote console without affecting any other console. + + * Added command "cdr show pgsql status" to check connection status + + * "sip show channel" will now display the complete route set. + + * Added "presencestate list" command. This command will list all custom + presence states that have been set by using the PRESENCE_STATE dialplan + function. + + * Added "presencestate change [,[,message[,options]]]" + command. This changes a custom presence to a new state. + + +Codecs +------------------- + * Codec lists may now be modified by the '!' character, to allow succinct + specification of a list of codecs allowed and disallowed, without the + requirement to use two different keywords. For example, to specify all + codecs except g729 and g723, one need only specify allow=all,!g729,!g723. + + * Add support for parsing SDP attributes, generating SDP attributes, and + passing it through. This support includes codecs such as H.263, H.264, SILK, + and CELT. You are able to set up a call and have attribute information pass. + This should help considerably with video calls. + + * The iLBC codec can now use a system-provided iLBC library if one is installed, + just like the GSM codec. + +DUNDi changes +------------- + * Added CLI commands dundi show hints and dundi show cache which will list DUNDi + 'DONTASK' hints in the cache and list all DUNDi cache entires respectively. + +Logging +------------------- + * Asterisk version and build information is now logged at the beginning of a + log file. + + * Threads belonging to a particular call are now linked with callids which get + added to any log messages produced by those threads. Log messages can now be + easily identified as involved with a certain call by looking at their call id. + Call ids may also be attached to log messages for just about any case where + it can be determined to be related to a particular call. + + * Each logging destination and console now have an independent notion of the + current verbosity level. Logger.conf now allows an optional argument to + the 'verbose' specifier, indicating the level of verbosity sent to that + particular logging destination. Additionally, remote consoles now each + have their own verbosity level. The command 'core set verbose' will now set + a separate level for each remote console without affecting any other + console. + + +Music On Hold +------------------- + * Added 'announcement' option which will play at the start of MOH and between + songs in modes of MOH that can detect transitions between songs (eg. + files, mp3, etc). + + +Parking +------------------- + * New per parking lot options: comebackcontext and comebackdialtime. See + configs/features.conf.sample for more details. + + * Channel variable PARKER is now set when comebacktoorigin is disabled in + a parking lot. + + * Channel variable PARKEDCALL is now set with the name of the parking lot + when a timeout occurs. + + +CDRs +------------------- + +CDR Postgresql Driver +------------------- + * Added command "cdr show pgsql status" to check connection status + + +CDR Adaptive ODBC Driver +------------------- + * Added schema option for databases that support specifying a schema. + + +Resource Modules +------------------- + +Calendars +------------------- + * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not + CALENDAR_WRITE has completed successfully. + + +res_rtp_asterisk +------------------- + * A new option, 'probation' has been added to rtp.conf + RTP in strictrtp mode can now require more than 1 packet to exit learning + mode with a new source (and by default requires 4). The probation option + allows the user to change the required number of packets in sequence to any + desired value. Use a value of 1 to essentially restore the old behavior. + Also, with strictrtp on, Asterisk will now drop all packets until learning + mode has successfully exited. These changes are based on how pjmedia handles + media sources and source changes. + + * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be + enabled or disabled using the icesupport setting. A variety of other + settings have been introduced to configure STUN/TURN connections. + + +res_corosync +------------------- + * A new module, res_corosync, has been introduced. This module uses the + Corosync cluster engineer (http://www.corosync.org) to allow a local cluster + of Asterisk servers to both Message Waiting Indication (MWI) and/or + Device State (presence) information. This module is very similar to, and + is a replacement for the res_ais module that was in previous releases of + Asterisk. + + +res_xmpp +------------------- + * This module adds a cleaned up, drop-in replacement for res_jabber called + res_xmpp. This provides the same externally facing functionality but is + implemented differently internally. res_jabber has been deprecated in favor + of res_xmpp; please see the UPGRADE.txt file for more information. + + +Scripts +------------------- + * The safe_asterisk script has been updated to allow several of its parameters + to be set from environment variables. This also enables a custom run + directory of Asterisk to be specified, instead of defaulting to /tmp. + + * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use + its value to determine the directory to assume is the top-level directory of + the source tree. If the variable is not set, it defaults to the current + behavior and uses the current working directory. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 1.8 to Asterisk 10 ------------------- +------------------------------------------------------------------------------ + +Text Messaging +-------------- + * Asterisk now has protocol independent support for processing text messages + outside of a call. Messages are routed through the Asterisk dialplan. + SIP MESSAGE and XMPP are currently supported. There are options in + jabber.conf and sip.conf to allow enabling these features. + -> jabber.conf: see the "sendtodialplan" and "context" options. + -> sip.conf: see the "accept_outofcall_message", "auth_message_requests" + and "outofcall_message_context" options. + The MESSAGE() dialplan function and MessageSend() application have been + added to go along with this functionality. More detailed usage information + can be found on the Asterisk wiki (http://wiki.asterisk.org/). + * If real-time text support (T.140) is negotiated, it will be preferred for + sending text via the SendText application. For example, via SIP, messages + that were once sent via the SIP MESSAGE request would be sent via RTP if + T.140 text is negotiated for a call. + +Parking +------- + * parkedmusicclass can now be set for non-default parking lots. + +Asterisk Manager Interface +-------------------------- + * PeerStatus now includes Address and Port. + * Added Hold events for when the remote party puts the call on and off hold + for chan_dahdi ISDN channels. + * Added new action MeetmeListRooms to list active conferences (shows same + data as "meetme list" at the CLI). + * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a + Description field that is set by 'description' in the channel configuration + file. + * Added Uniqueid header to UserEvent. + * Added new action FilterAdd to control event filters for the current session. + This requires the system permission and uses the same filter syntax as + filters that can be defined in manager.conf + * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous + versions had some instances of the event converted, but others were left + as-is. All Unlink events should now be converted to Bridge events. The AMI + protocol version number was incremented to 1.2 as a result of this change. + +Asterisk HTTP Server +-------------------------- + * The HTTP Server can bind to IPv6 addresses. + +chan_dahdi +-------------------------- + * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used + with busydetect. usage example: busypattern=200,200,200,600 + +CLI Changes +-------------------------- + * New 'gtalk show settings' command showing the current settings loaded from + gtalk.conf. + * The 'logger reload' command now supports an optional argument, specifying an + alternate configuration file to use. + * 'dialplan add extension' command will now automatically create a context if + the specified context does not exist with a message indicated it did so. + * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a + Description field which can be populated with 'description' in the channel + configuration files (sip.conf, iax2.conf, and chan_dahdi.conf). + +CDR +-------------------------- + * The filter option in cdr_adaptive_odbc now supports negating the argument, + thus allowing records which do NOT match the specified filter. + * Added ability to log CONGESTION calls to CDR + +CODECS +-------------------------- + * Ability to define custom SILK formats in codecs.conf. + * Addition of speex32 audio format with translation. + * CELT codec pass-through support and ability to define + custom CELT formats in codecs.conf. + * Ability to read raw signed linear files with sample rates + ranging from 8khz - 192khz. The new file extensions introduced + are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192. + * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP, + Skinny, H.323, etc) can still only support the following codecs: + Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm, + siren7, siren14, speex, speex16, ilbc, lpc10, adpcm + Video: h261, h263, h263p, h264, mpeg4 + Image: jpeg, png + Text: red, t140 + +ConfBridge +-------------------------- + * New highly optimized and customizable ConfBridge application capable of + mixing audio at sample rates ranging from 8khz-96khz. + * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user + and bridge profiles on a channel. + * CONFBRIDGE_INFO dialplan function capable of retrieving information + about a conference such as locked status and number of parties, admins, + and marked users. + * Addition of video_mode option in confbridge.conf for adding video support + into a bridge profile. + * Addition of the follow_talker video_mode in confbridge.conf. This video + mode dynamically switches the video feed to always display the loudest talker + supplying video in the conference. + +Dialplan Variables +------------------ + * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR, + ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent + variables from asterisk.conf. + +Dialplan Functions +------------------ + * Addition of the JITTERBUFFER dialplan function. This function allows + for jitterbuffering to occur on the read side of a channel. By using + this function conference applications such as ConfBridge and MeetMe can + have the rx streams jitterbuffered before conference mixing occurs. + * Added DB_KEYS, which lists the next set of keys in the Asterisk database + hierarchy. + * Added STRREPLACE function. This function let's the user search a variable + for a given string to replace with another string as many times as the + user specifies or just throughout the whole string. + * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel. + * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS() + * Added extensions to chan_ooh323 in function CHANNEL() + +libpri channel driver (chan_dahdi) DAHDI changes +-------------------------- + * Added moh_signaling option to specify what to do when the channel's bridged + peer puts the ISDN channel on hold. + * Added display_send and display_receive options to control how the display ie + is handled. To send display text from the dialplan use the SendText() + application when the option is enabled. + * Added mcid_send option to allow sending a MCID request on a span. + +Calendaring +-------------------------- + * Added setvar option to calendar.conf to allow setting channel variables on + notification channels. + * Added "calendar show types" CLI command to list registered calendar + connectors. + +MixMonitor +-------------------------- + * Added two new options, r and t with file name arguments to record + single direction (unmixed) audio recording separate from the bidirectional + (mixed) recording. The mixed file name argument is optional now as long + as at least one recording option is used. + +FollowMe +-------------------------- + * Added a new option, l, which will disable local call optimization for + channels involved with the FollowMe thread. Use this option to improve + compatability for a FollowMe call with certain dialplan apps, options, and + functions. + +Meetme +-------------------------- + * Added option "k" that will automatically close the conference when there's + only one person left when a user exits the conference. + +CEL +-------------------------- + * cel_pgsql now supports the 'extra' column for data added using the + CELGenUserEvent() application. + +pbx_lua +-------------------------- + * Support for defining hints has been added to pbx_lua. See the 'hints' table + in the sample extensions.lua file for syntax details. + * Applications that perform jumps in the dialplan such as Goto will now + execute properly. When pbx_lua detects that the context, extension, or + priority we are executing on has changed it will immediately return control + to the asterisk PBX engine. Currently the engine cannot detect a Goto to + the priority after the currently executing priority. + * An autoservice is now started by default for pbx_lua channels. It can be + stopped and restarted using the autoservice_stop() and autoservice_start() + functions. + +res_fax +-------------------------- + * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated + into a FAXStatus event with an 'Operation' header that will be either + 'send', 'receive', and 'gateway'. + * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp). + Set FAXOPT(gateway)=yes to enable this functionality on a channel. This + feature will handle converting a fax call between an audio T.30 fax terminal + and an IFP T.38 fax terminal. + +SIP Changes +----------- + * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected. + * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently. + * SIP now generates security events using the Security Events Framework for REGISTER and INVITE. + +Queue changes +------------- + * Added general option negative_penalty_invalid default off. when set + members are seen as invalid/logged out when there penalty is negative. + for realtime members when set remove from queue will set penalty to -1. + * Added queue option autopausedelay when autopause is enabled it will be + delayed for this number of seconds since last successful call if there + was no prior call the agent will be autopaused immediately. + * Added member option ignorebusy this when set and ringinuse is not + will allow per member control of multiple calls as ringinuse does for + the Queue. + +Applications +------------ + * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves + a MeetMe conference + * Added 'k' option to MeetMe to automatically kill the conference when there's only + one participant left (much like a normal call bridge) + * Added extra argument to Originate to set timeout. + +Asterisk Database +----------------- + * The internal Asterisk database has been switched from Berkeley DB 1.86 to + SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3 + utility in the UTILS section of menuselect. If an existing astdb is found and no + astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will + convert an existing astdb to the SQLite3 version automatically at runtime. + +Asterisk Modules +---------------- + * Modules marked as deprecated are no longer marked as building by default. Enabling + these modules is still available via menuselect. + +IAX2 Changes +------------ + * authdebug is now disabled by default. To enable this functionality again + set authdebug = yes in iax.conf. + +RTP Changes +----------- + * The rtp.conf setting "strictrtp" is now enabled by default. In previous + releases it was disabled. + +PBX Core +-------- + * The PBX core previously made a call with a non-existing extension test for + extension s@default and jump there if the extension existed. + This was a bad default behaviour and violated the principle of least surprise. + It has therefore been changed in this release. It may affect some + applications and configurations that rely on this behaviour. Most channel + drivers have avoided this for many releases by testing whether the extension + called exists before starting the PBX and generating a local error. + This behaviour still exists and works as before. + + Extension "s" is used when no extension is given in a channel driver, + like immediate answer in DAHDI or calling to a domain with no user part + in a SIP uri. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ---------------- +------------------------------------------------------------------------------ + +SIP Changes +----------- + * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf + now defaults to force_rport. It is very important that phones requiring nat=no be + specifically set as such instead of relying on the default setting. If at all + possible, all devices should have nat settings configured in the general section as + opposed to configuring nat per-device. + * Added preferred_codec_only option in sip.conf. This feature limits the joint + codecs sent in response to an INVITE to the single most preferred codec. + * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec + to be used for the outgoing call. It must be one of the codecs configured + for the device. + * Added tlsprivatekey option to sip.conf. This allows a separate .pem file + to be used for holding a private key. If tlsprivatekey is not specified, + tlscertfile is searched for both public and private key. + * Added tlsclientmethod option to sip.conf. This allows the protocol for + outbound client connections to be specified. + * The sendrpid parameter has been expanded to include the options + 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID + header to be sent (equivalent to setting sendrpid=yes) and setting + sendrpid to 'pai' will cause P-Asserted-Identity header to be sent. + * The 'ignoresdpversion' behavior has been made automatic when the SDP received + is in response to a T.38 re-INVITE that Asterisk initiated. In this situation, + since the call will fail if Asterisk does not process the incoming SDP, Asterisk + will accept the SDP even if the SDP version number is not properly incremented, + but will generate a warning in the log indicating that the SIP peer that sent + the SDP should have the 'ignoresdpversion' option set. + * The 'nat' option has now been been changed to have yes, no, force_rport, and + comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables + symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the + remote side requests it and disables symmetric RTP support. Setting it to + force_rport forces RFC 3581 behavior and disables symmetric RTP support. + Setting it to comedia enables RFC 3581 behavior if the remote side requests it + and enables symmetric RTP support. + * Slave SIP channels now set HASH(SIP_CAUSE,) on each + response. This permits the master channel to know how each channel dialled + in a multi-channel setup resolved in an individual way. This carries a + performance penalty and can be disabled in sip.conf using the + 'storesipcause' option. + * Added 'externtcpport' and 'externtlsport' options to allow custom port + configuration for the externip and externhost options when tcp or tls is used. + * Added support for message body (stored in content variable) to SIP NOTIFY message + accessible via AMI and CLI. + * Added 'media_address' configuration option which can be used to explicitly specify + the IP address to use in the SDP for media (audio, video, and text) streams. + * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox + that the new/old count should be stored on if an unsolicited MWI NOTIFY message is + received. + * Added 'use_q850_reason' configuration option for generating and parsing + if available Reason: Q.850;cause= header. It is implemented + in some gateways for better passing PRI/SS7 cause codes via SIP. + * When dialing SIP peers, a new component may be added to the end of the dialstring + to indicate that a specific remote IP address or host should be used when dialing + the particular peer. The dialstring format is SIP/peer/exten/host_or_IP. + * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The + ability to selectively force bridged channels to also be encrypted is also + implemented. Branching in the dialplan can be done based on whether or not + a channel has secure media and/or signaling. + * Added directmediapermit/directmediadeny to limit which peers can send direct media + to each other + * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of + Charge messages to snom phones. + * Added support for G.719 media streams. + * Added support for 16khz signed linear media streams. + * SIP is now able to bind to and communicate with IPv6 addresses. In addition, + RTP has been outfitted with the same abilities. + * Added support for setting the Max-Forwards: header in SIP requests. Setting is + available in device configurations as well as in the dial plan. + * Addition of the 'subscribe_network_change' option for turning on and off + res_stun_monitor module support in chan_sip. + * Addition of the 'auth_options_requests' option for turning on and off + authentication for OPTIONS requests in chan_sip. + +Configuration files +------------------- + * Add #tryinclude statement for config files. This provides the same + functionality as the #include statement however an asterisk module will + still load if the filename does not exist. Using the #include statement + Asterisk will not allow the module to load. + +IAX2 Changes +----------- + * Added rtsavesysname option into iax.conf to allow the systname to be saved + on realtime updates. + * Added the ability for chan_iax2 to inform the dialplan whether or not + encryption is being used. This interoperates with the SIP SRTP implementation + so that a secure SIP call can be bridged to a secure IAX call when the + dialplan requires bridged channels to be "secure". + * Addition of the 'subscribe_network_change' option for turning on and off + res_stun_monitor module support in chan_iax. + + +MGCP Changes +------------ + * Added ability to preset channel variables on indicated lines with the setvar + configuration option. Also, clearvars=all resets the list of variables back + to none. + * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks. + See configs/res_pktccops.conf for more information. + +XMPP Google Talk/Jingle changes +------------------------------- + * Added the externip option to gtalk.conf. + * Added the stunaddr option to gtalk.conf which allows for the automatic + retrieval of the external ip from a stun server. + +Applications +------------ + * Added 'p' option to PickupChan() to allow for picking up channel by the first + match to a partial channel name. + * Added .m3u support for Mp3Player application. + * Added progress option to the app_dial D() option. When progress DTMF is + present, those values are sent immediately upon receiving a PROGRESS message + regardless if the call has been answered or not. + * Added functionality to the app_dial F() option to continue with execution + at the current location when no parameters are provided. + * Added the 'a' option to app_dial to answer the calling channel before any + announcements or macros are executed. + * Modified app_dial to set answertime when the called channel answers even if + the called channel hangs up during playback of an announcement. + * Modified app_dial 'r' option to support an additional parameter to play an + indication tone from indications.conf + * Added c() option to app_chanspy. This option allows custom DTMF to be set + to cycle through the next available channel. By default this is still '*'. + * Added x() option to app_chanspy. This option allows DTMF to be set to + exit the application. + * The Voicemail application has been improved to automatically ignore messages + that only contain silence. + * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the + associated mailbox(es) to be greetings-only. + * The ChanSpy application now has the 'S' option, which makes the application + automatically exit once it hits a point where no more channels are available + to spy on. + * The ChanSpy application also now has the 'E' option, which spies on a single + channel and exits when that channel hangs up. + * The MeetMe application now turns on the DENOISE() function by default, for + each participant. In our tests, this has significantly decreased background + noise (especially noisy data centers). + * Voicemail now permits storage of secrets in a separate file, located in the + spool directory of each individual user. The control for this is located in + the "passwordlocation" option in voicemail.conf. Please see the sample + configuration for more information. + * The ChanIsAvail application now exposes the returned cause code using a separate + variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS. + * Added 'd' option to app_followme. This option disables the "Please hold" + announcement. + * Added 'y' option to app_record. This option enables a mode where any DTMF digit + received will terminate recording. + * Voicemail now supports per mailbox settings for folders when using IMAP storage. + Previously the folder could only be set per context, but has now been extended + using the imapfolder option. + * Voicemail now supports per mailbox settings for nextaftercmd and minsecs. + * Voicemail now allows the pager date format to be specified separately from the + email date format. + * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added + to allow joining, leaving, and sending text to group chats. + * MeetMe has a new option 'G' to play an announcement before joining a conference. + * Page has a new option 'A(x)' which will playback an announcement simultaneously + to all paged phones (and optionally excluding the caller's one using the new + option 'n') before the call is bridged. + * The 'f' option to Dial has been augmented to take an optional argument. If no + argument is provided, the 'f' option works as it always has. If an argument is + provided, then the connected party information of all outgoing channels created + during the Dial will be set to the argument passed to the 'f' option. + * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a + Gosub on the peer. + * The OSP lookup application adds in/outbound network ID, optional security, + number portability, QoS reporting, destination IP port, custom info and service + type features. + * Added new application VMSayName that will play the recorded name of the voicemail + user if it exists, otherwise will play the mailbox number. + * Added custom device states to ConfBridge bridges. Use 'confbridge:' to + retrieve state for a particular bridge, where is the conference name + * app_directory now allows exiting at any time using the operator or pound key. + * Voicemail now supports setting a locale per-mailbox. + * Two new applications are provided for declining counting phrases in multiple + languages. See the application notes for SayCountedNoun and SayCountedAdj for + more information. + * Voicemail now runs the externnotify script when pollmailboxes is activated and + notices a change. + * Voicemail now includes rdnis within msgXXXX.txt file. + * ExternalIVR now supports IPv6 addresses. + * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki + at https://wiki.asterisk.org/wiki/x/oQBB + * ParkedCall and Park can now specify the parking lot to use. + +Dialplan Functions +------------------ + * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate + over SRV records associated with a specific service. From the CLI, type + 'core show function SRVQUERY' and 'core show function SRVRESULT' for more + details on how these may be used. + * PITCH_SHIFT dialplan function added. This function can be used to modify the + pitch of a channel's tx and rx audio streams. + * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits + setting various connected line and redirecting party information. + * CALLERID and CONNECTEDLINE dialplan functions have been extended to + support ISDN subaddressing. + * The CHANNEL() function now supports the "name" and "checkhangup" options. + * For DAHDI channels, the CHANNEL() dialplan function now allows + the dialplan to request changes in the configuration of the active + echo canceller on the channel (if any), for the current call only. + The syntax is: + + exten => s,n,Set(CHANNEL(echocan_mode)=off) + + The possible values are: + + on - normal mode (the echo canceller is actually reinitialized) + off - disabled + fax - FAX/data mode (NLP disabled if possible, otherwise completely + disabled) + voice - voice mode (returns from FAX mode, reverting the changes that + were made when FAX mode was requested) + * Added new dialplan function MASTER_CHANNEL(), which permits retrieving + and setting variables on the channel which created the current channel. + Administrators should take care to avoid naming conflicts, when multiple + channels are dialled at once, especially when used with the Local channel + construct (which all could set variables on the master channel). Usage + of the HASH() dialplan function, with the key set to the name of the slave + channel, is one approach that will avoid conflicts. + * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound + audio in a channel. + * func_odbc now allows multiple row results to be retrieved without using + mode=multirow. If rowlimit is set, then additional rows may be retrieved + from the same query by using the name of the function which retrieved the + first row as an argument to ODBC_FETCH(). + * Added JABBER_RECEIVE, which permits receiving XMPP messages from the + dialplan. This function returns the content of the received message. + * Added REPLACE, which searches a given variable name for a set of characters, + then either replaces them with a single character or deletes them. + * Added PASSTHRU, which literally passes the same argument back as its return + value. The intent is to be able to use a literal string argument to + functions that currently require a variable name as an argument. + * HASH-associated variables now can be inherited across channel creation, by + prefixing the name of the hash at assignment with the appropriate number of + underscores, just like variables. + * GROUP_MATCH_COUNT has been improved to allow regex matching on category + * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get + whether or not channels that are bridged to the current channel will be + required to have secure signaling and/or media. + * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not + the current channel has secure signaling and/or media. + * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the + "no_media_path" option. + Returns "0" if there is a B channel associated with the call. + Returns "1" if no B channel is associated with the call. The call is either + on hold or is a call waiting call. + * Added option to dialplan function CDR(), the 'f' option + allows for high resolution times for billsec and duration fields. + * FILE() now supports line-mode and writing. + * Added FIELDNUM(), which returns the 1-based offset of a field in a list. + * FRAME_TRACE(), for tracking internal ast_frames on a channel. + +Dialplan Variables +------------------ + * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature. + * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side + and is set when a dynamic feature is triggered. + * Added PARKINGLOT which can be used with parkeddynamic feature.conf option + to dynamically create a new parking lot matching the value this varible is + set to. + * Added PARKINGDYNAMIC which represents the template parkinglot defined in + features.conf that should be the base for dynamic parkinglots. + * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic + parkinglot should have. + * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic + parkinglot should have. + * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot + should have. + +Queue changes +------------- + * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up + timeout has expired. + * Added 'R' option to app_queue. This option stops moh and indicates ringing + to the caller when an Agent's phone is ringing. This can be used to indicate + to the caller that their call is about to be picked up, which is nice when + one has been on hold for an extened period of time. + * A new config option, penaltymemberslimit, has been added to queues.conf. + When set this option will disregard penalty settings when a queue has too + few members. + * A new option, 'I' has been added to both app_queue and app_dial. + By setting this option, Asterisk will not update the caller with + connected line changes or redirecting party changes when they occur. + * A 'relative-periodic-announce' option has been added to queues.conf. When + enabled, this option will cause periodic announce times to be calculated + from the end of announcements rather than from the beginning. + * The autopause option in queues.conf can be passed a new value, "all." The + result is that if a member becomes auto-paused, he will be paused in all + queues for which he is a member, not just the queue that failed to reach + the member. + * Added dialplan function QUEUE_EXISTS to check if a queue exists + * The queue logger now allows events to optionally propagate to a file, + even when realtime logging is turned on. Additionally, realtime logging + supports sending the event arguments to 5 individual fields, although it + will fallback to the previous data definition, if the new table layout is + not found. + +mISDN channel driver (chan_misdn) changes +---------------------------------------- + * Added display_connected parameter to misdn.conf to put a display string + in the CONNECT message containing the connected name and/or number if + the presentation setting permits it. + * Added display_setup parameter to misdn.conf to put a display string + in the SETUP message containing the caller name and/or number if the + presentation setting permits it. + * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to + indicate the dialplan settings are to be obtained from the asterisk + channel. + * Made misdn.conf parameter callerid accept the "name" format + used by the rest of the system. + * Made use the nationalprefix and internationalprefix misdn.conf + parameters to prefix any received number from the ISDN link if that + number has the corresponding Type-Of-Number. NOTE: This includes + comparing the incoming call's dialed number against the MSN list. + * Added the following new parameters: unknownprefix, netspecificprefix, + subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any + received number from the ISDN link if that number has the corresponding + Type-Of-Number. + * Added new dialplan application misdn_command which permits controlling + the CCBS/CCNR functionality. + * Added new dialplan function mISDN_CC which permits retrieval of various + values from an active call completion record. + * For PTP, you should manually send the COLR of the redirected-to party + for an incomming redirected call if the incoming call could experience + further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and + set the REDIRECTING(to-pres) to the COLR. A call has been redirected + if the REDIRECTING(from-num) is not empty. + * For outgoing PTP redirected calls, you now need to use the inhibit(i) + option on all of the REDIRECTING statements before dialing the + redirected-to party. You still have to set the REDIRECTING(to-xxx,i) + and the REDIRECTING(from-xxx,i) values. The PTP call will update the + redirecting-to presentation (COLR) when it becomes available. + * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP + information. + +thirdparty mISDN enhancements +----------------------------- +mISDN has been modified by Digium, Inc. to greatly expand facility message +support to allow: + * Enhanced COLP support for call diversion and transfer. + * CCBS/CCNR support. + +The latest modified mISDN v1.1.x based version is available at: +http://svn.digium.com/svn/thirdparty/mISDN/trunk +http://svn.digium.com/svn/thirdparty/mISDNuser/trunk + +Tagged versions of the modified mISDN code are available under: +http://svn.digium.com/svn/thirdparty/mISDN/tags +http://svn.digium.com/svn/thirdparty/mISDNuser/tags + +libpri channel driver (chan_dahdi) DAHDI changes +------------------------------------------- + * The channel variable PRIREDIRECTREASON is now just a status variable + and it is also deprecated. Use the REDIRECTING(reason) dialplan function + to read and alter the reason. + * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the + redirected-to party for an incomming redirected call if the incoming call + could experience further redirects. Just set the + REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres) + to the COLR. A call has been redirected if the REDIRECTING(count) is not + zero. + * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to + use the inhibit(i) option on all of the REDIRECTING statements before + dialing the redirected-to party. You still have to set the + REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call + will update the redirecting-to presentation (COLR) when it becomes available. + * Added the ability to ignore calls that are not in a Multiple Subscriber + Number (MSN) list for PTMP CPE interfaces. + * Added dynamic range compression support for dahdi channels. It is + configured via the rxdrc and txdrc parameters in chan_dahdi.conf. + * Added support for ISDN calling and called subaddress with partial support + for connected line subaddress. + * Added support for BRI PTMP NT mode. (Requires latest LibPRI.) + * Added handling of received HOLD/RETRIEVE messages and the optional ability + to transfer a held call on disconnect similar to an analog phone. + * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. + Will reroute/deflect an outgoing call when receive the message. + Can use the DAHDISendCallreroutingFacility to send the message for the + supported switches. + * Added standard location to add options to chan_dahdi dialing: + Dial(DAHDI/g1[/extension[/options]]) + Current options: + K() + R Reverse charging indication + * Added Reverse Charging Indication (Collect calls) send/receive option. + Send reverse charging in SETUP message with the chan_dahdi R dialing option. + Dial(DAHDI/g1/extension/R) + Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)} + (requires latest LibPRI) + * Added ability to send/receive keypad digits in the SETUP message. + Send keypad digits in SETUP message with the chan_dahdi K() + dialing option. Dial(DAHDI/g1/[extension]/K()) + Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} + (requires latest LibPRI) + * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages + to eliminate tromboned calls. A tromboned call goes out an interface and comes + back into the same interface. Tromboned calls happen because of call routing, + call deflection, call forwarding, and call transfer. + * Added the ability to send and receive ETSI Advice-Of-Charge messages. + * Added the ability to support call waiting calls. (The SETUP has no B channel + assigned.) + * Added Malicious Call ID (MCID) event to the AMI call event class. + * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones). + +Asterisk Manager Interface +-------------------------- + * The Hangup action now accepts a Cause header which may be used to + set the channel's hangup cause. + * sslprivatekey option added to manager.conf and http.conf. Adds the ability + to specify a separate .pem file to hold a private key. By default sslcert + is used to hold both the public and private key. + * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced + for options containing the 'tls' prefix. For example, 'sslenable' is now + 'tlsenable'. This has been done in effort to keep ssl and tls options consistent + across all .conf files. All affected sample.conf files have been modified to + reflect this change. Previous options such as 'sslenable' still work, + but options with the 'tls' prefix are preferred. + * Added a MuteAudio AMI action for muting inbound and/or outbound audio + in a channel. (res_mutestream.so) + * The configuration file manager.conf now supports a channelvars option, which + specifies a list of channel variables to include in each channel-oriented + event. + * The redirect command now has new parameters ExtraContext, ExtraExtension, + and ExtraPriority to allow redirecting the second channel to a different + location than the first. + * Added new event "JabberStatus" in the Jabber module to monitor buddies + status. + * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio + in a MixMonitor recording. + * The 'iax2 show peers' output is now similar to the expected output of + 'sip show peers'. + * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new + aoc event class. + * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and + AOC-E messages on a channel. + * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action + conform more closely to similar events. + * Added a new eventfilter option per user to allow whitelisting and blacklisting + of events. + * Added optional parkinglot variable for park command. + * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses + if CallerIDNum and CallerIDName headers are also present. + +Channel Event Logging +--------------------- + * A new interface, CEL, is introduced here. CEL logs single events, much like + the AMI, but it differs from the AMI in that it logs to db backends much + like CDR does; is based on the event subsystem introduced by Russell, and + can share in all its benefits; allows multiple backends to operate like CDR; + is specialized to event data that would be of concern to billing systems, + like CDR. Backends for logging and accounting calls have been produced, + but a new CDR backend is still in development. + +CDR +--- + * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados. + linkedid is based on uniqueID, but spreads to other channels as transfers, dials, + etc are performed. Thus the pieces of CDR can be grouped into multilegged sets. + * Multiple files and formats can now be specified in cdr_custom.conf. + * cdr_syslog has been added which allows CDRs to be written directly to syslog. + See configs/cdr_syslog.conf.sample for more information. + * A 'sequence' field has been added to CDRs which can be combined with + linkedid or uniqueid to uniquely identify a CDR. + * Handling of billsec and duration field has changed. If your table definition + specifies those fields as float,double or similar they will now be logged with + microsecond accuracy instead of a whole integer. + +Calendaring for Asterisk +------------------------ + * A new set of modules were added supporting calendar integration with Asterisk. + Dialplan functions for reading from and writing to calendars are included, + as well as the ability to execute dialplan logic upon calendar event notifications. + iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for + Exchange Server 2003 with no write or attendee support, and res_calendar_ews for + Exchange Server 2007+ with full write and attendee support) are supported (Exchange + 2003 support does not support forms-based authentication). + +Call Completion Supplementary Services for Asterisk +--------------------------------------------------- + * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog. + DAHDI/ISDN supports call completion for the following switch types: + EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig. + See https://wiki.asterisk.org/wiki/x/2ABQ for details. + +Multicast RTP Support +--------------------- + * A new RTP engine and channel driver have been added which supports Multicast RTP. + The channel driver can be used with the Page application to perform multicast RTP + paging. The dial string format is: MulticastRTP/// + Type can be either basic or linksys. + Destination is the IP address and port for the RTP packets. + Control address is specific to the linksys type and is used for sending the control + packets unique to them. + +Security Events Framework +------------------------- + * Asterisk has a new C API for reporting security events. The module res_security_log + sends these events to the "security" logger level. Currently, AMI is the only + Asterisk component that reports security events. However, SIP support will be + coming soon. For more information on the security events framework, see the + "Asterisk Security Framework" section of the Asterisk wiki at + https://wiki.asterisk.org/wiki/x/wgBQ + * SIP support was added in Asterisk 10 + * This API now supports IPv6 addresses + +Fax +--- + * A technology independent fax frontend (res_fax) has been added to Asterisk. + * A spandsp based fax backend (res_fax_spandsp) has been added. + * The app_fax module has been deprecated in favor of the res_fax module and + the new res_fax_spandsp backend. + * The SendFAX and ReceiveFAX applications now send their log messages to a + 'fax' logger level, instead of to the generic logger levels. To see these + messages, the system's logger.conf file will need to direct the 'fax' logger + level to one or more destinations; the logger.conf.sample file includes an + example of how to do this. Note that if the 'fax' logger level is *not* + directed to at least one destination, log messages generated by these + applications will be lost, and that if the 'fax' logger level is directed to + the console, the 'core set verbose' and 'core set debug' CLI commands will + have no effect on whether the messages appear on the console or not. + +Miscellaneous +------------- + * The transmit_silence_during_record option in asterisk.conf.sample has been removed. + Now, in order to enable transmitting silence during record the transmit_silence + option should be used. transmit_silence_during_record remains a valid option, but + defaults to the behavior of the transmit_silence option. + * Addition of the Unit Test Framework API for managing registration and execution + of unit tests with the purpose of verifying the operation of C functions. + * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send + XMPP text messages to the remote JID. + * Modules.conf has a new option - "require" - that marks a module as critical for + the execution of Asterisk. + If one of the required modules fail to load, Asterisk will exit with a return + code set to 2. + * An 'X' option has been added to the asterisk application which enables #exec support. + This allows #exec to be used in asterisk.conf. + * jabber.conf supports a new option auth_policy that toggles auto user registration. + * A new lockconfdir option has been added to asterisk.conf to protect the + configuration directory (/etc/asterisk by default) during reloads. + * The parkeddynamic option has been added to features.conf to enable the creation + of dynamic parkinglots. + * chan_dahdi now supports reporting alarms over AMI either by channel or span via + the reportalarms config option. + * chan_dahdi supports dialing configuring and dialing by device file name. + DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise + it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. + * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. + False by default. If set, chan_dahdi will ignore failed 'channel' entries. + Handy for the above name-based syntax as it does not depend on + initialization order. + * The Realtime dialplan switch now caches entries for 1 second. This provides a + significant increase in performance (about 3X) for installations using this switchtype. + * Distributed devicestate now supports the use of the XMPP protocol, in addition to + AIS. For more information, please see the Distributed Device State section of the + Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ + * The addition of G.719 pass-through support. + * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16' + during device configuration. + * The UNISTIM channel driver (chan_unistim) has been updated to support devices that + have less than 3 lines on the LCD. + * Realtime now supports database failover. See the sample extconfig.conf for details. + * The addition of improved translation path building for wideband codecs. Sample + rate changes during translation are now avoided unless absolutely necessary. + * The addition of the res_stun_monitor module for monitoring and reacting to network + changes while behind a NAT. + * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf. + DTMF Valid/Invalid number of hits/misses can be set in dsp.conf. + These allow support for any Administration. Default is AT&T values. + +CLI Changes +----------- + * The 'core set debug' and 'core set verbose' commands, in previous versions, could + optionally accept a filename, to apply the setting only to the code generated from + that source file when Asterisk was built. However, there are some modules in Asterisk + that are composed of multiple source files, so this did not result in the behavior + that users expected. In this version, 'core set debug' and 'core set verbose' + can optionally accept *module* names instead (with or without the .so extension), + which applies the setting to the entire module specified, regardless of which source + files it was built from. + * New 'manager show settings' command showing the current settings loaded from + manager.conf. + * Added 'all' keyword to the CLI command "channel request hangup" so that you can send + the channel hangup request to all channels. + * Added a "core reload" CLI command that executes a global reload of Asterisk. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 ------------- +------------------------------------------------------------------------------ + +SIP Changes +----------- + * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups. + Snom phones use this for call pickup of extensions that the phone is + subscribed to. + * Added support for setting the domain in the URI for caller of an + outbound call by using the SIPFROMDOMAIN channel variable. + * Added a new configuration option "remotesecret" for authentication to + remote services. For backwards compatibility, "secret" still has the + same function as before, but now you can configure both a remote secret and a + local secret for mutual authentication. + * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set, + the sound will be played to the target of an attended transfer + * Added two new configuration options, "qualifygap" and "qualifypeers", which allow + finer control over how many peers Asterisk will qualify and the gap between them + when all peers need to be qualified at the same time. + * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled + (either globally or for a specific peer), chan_sip will treat any SDP data + it receives as new data and update the media stream accordingly. By + default, Asterisk will only modify the media stream if the SDP session + version received is different from the current SDP session version. This + option is required to interoperate with devices that have non-standard SDP + session version implementations (observed with Microsoft OCS). This option + is disabled by default. + * The parsing of register => lines in sip.conf has been modified to allow a port + to be present in the "user" portion. Please see the sip.conf.sample file for more + information + * Added support for subscribing to MWI on a remote server and making the status available + as a mailbox. Please see the sip.conf.sample file for more information. + * Added a function to remove SIP headers added in the dialplan before the + first INVITE is generated - SIPRemoveHeader() + * Channel variables set with setvar= in a device configuration is now + set both for inbound and outbound calls. + * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams. + +IAX2 changes +------------ + * Added immediate option to iax.conf + * Added forceencryption option to iax.conf + * Added Encryption and Trunk status to manager command "iaxpeers" + +Skinny Changes +-------------- + * The configuration file now holds separate sections for devices and lines. + Please have a look at configs/skinny.conf.sample and change your skinny.conf + accordingly. + +DAHDI Changes +------------- + * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with + support for LibOpenR2. http://www.libopenr2.org/ + * The UK option waitfordialtone has been added for use with BT analog + lines. + * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option + is used in conjunction with the 'faxdetect' configuration option. When + 'faxbuffers' is used and fax tones are detected, the channel will dynamically + switch to the configured faxbuffers policy. For example, to use 6 buffers + and a 'full' buffer policy for a fax transmission, add: + faxbuffers=>6,full + The faxbuffers configuration will be in affect until the call is torn down. + * Added service message support for 4ESS/5ESS switches. + +Dialplan Functions +------------------ + * For DAHDI channels, the CHANNEL() dialplan function now + supports changing the channel's buffer policy (for the current + call only), using this syntax: + + exten => s,n,Set(CHANNEL(buffers)=6,full) + + This would change the channel to the 'full' buffer policy and + 6 (six) buffers. Possible options for this setting are the same + as those in chan_dahdi.conf. + * Added a new dialplan function, CURLOPT, which permits setting various + options that may be useful with the CURL dialplan function, such as + cookies, proxies, connection timeouts, passwords, etc. + * Permit the syntax and synopsis fields of the corresponding dialplan + functions to be individually set from func_odbc.conf. + * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'. + * func_odbc now may specify an insert query to execute, when the write query + affects 0 rows (usually indicating that no such row exists). + * Added a new dialplan function, LISTFILTER, which permits removing elements + from a set list, by name. Uses the same general syntax as the existing CUT + and FIELDQTY dialplan functions, which also manage lists. + * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better + obtaining realtime data from the dialplan. + * Added LOCAL_PEEK, which allows access to variables in any stack frame within + a subroutine when using the GoSub() and Return() applications. + * Added AUDIOHOOK_INHERIT. For information on its use, please see the output + of "core show function AUDIOHOOK_INHERIT" from the CLI + * Added AES_ENCRYPT. For information on its use, please see the output + of "core show function AES_ENCRYPT" from the CLI + * Added AES_DECRYPT. For information on its use, please see the output + of "core show function AES_DECRYPT" from the CLI + * func_odbc now supports database transactions across multiple queries. + +Applications +------------ + * Scheduled meetme conferences may now have their end times extended by + using MeetMeAdmin. + * app_authenticate now gives the ability to select a prompt other than + the default. + * app_directory now pays attention to the searchcontexts setting in + voicemail.conf and will look through all contexts, if no context is + specified in the initial argument. + * A new application, Originate, has been introduced, that allows asynchronous + call origination from the dialplan. + * Voicemail now permits setting the emailsubject and emailbody per mailbox, + in addition to the setting in the "general" context. + * Added ConfBridge dialplan application which does conference bridges without + DAHDI. For information on its use, please see the output of + "core show application ConfBridge" from the CLI. + +Miscellaneous +------------- + * The Asterisk CLI has a new command, "channel redirect", which is similar in + operation to the AMI Redirect action. + * extensions.conf now allows you to use keyword "same" to define an extension + without actually specifying an extension. It uses exactly the same pattern + as previously used on the last "exten" line. For example: + exten => 123,1,NoOp(something) + same => n,SomethingElse() + * musiconhold.conf classes of type 'files' can now use relative directory paths, + which are interpreted as relative to the astvarlibdir setting in asterisk.conf. + * All deprecated CLI commands are removed from the sourcecode. They are now handled + by the new clialiases module. See cli_aliases.conf.sample file. + * Times within timespecs are now accurate down to the minute. This is a change + from historical Asterisk, which only provided timespecs rounded to the nearest + even (read: evenly divisible by 2) minute mark. + * The realtime switch now supports an option flag, 'p', which disables searches for + pattern matches. + * In addition to a time range and date range, timespecs now accept a 5th optional + argument, timezone. This allows you to perform time checks on alternate + timezones, especially if those daylight savings time ranges vary from your + machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed + includes. + * The contrib/scripts/ directory now has a script called sip_nat_settings that will + give you the correct output for an asterisk box behind nat. It will give you the + externhost and localnet settings. + * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and + can connect calls in passthrough mode, as well as record and play back files. + * Successful and unsuccessful call pickup can now be alerted through sounds, by + using pickupsound and pickupfailsound in features.conf. + * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default. + This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX + instead of the /var/run/asterisk.pid where it used to be. This will make + installs as non-root easier to manage. + +CDR +--- + +* The cdr.conf file must exist and be correctly programmed in order for CDR records to + be written; they will no longer be explicitly written. + +Asterisk Manager Interface +-------------------------- + * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with + a non-empty value) in your request. If you do this, any pending AMI events will + *not* be included in the response to your request as they would normally, but + will be left in the event queue for the next request you make to retrieve. For + some applications, this will allow you to guarantee that you will only see + events in responses to 'WaitEvent' actions, and can better know when to expect them. + To know whether the Asterisk server supports this header or not, your client can + inspect the first response back from the server to see if it includes this header: + + Pragma: SuppressEvents + + If this is included, the server supports event suppression. + + * Added 4 new Actions to list skinny device(s) and line(s) + SKINNYdevices + SKINNYshowdevice + SKINNYlines + SKINNYshowline + +LDAP Schema File Additions +-------------------------- + * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses + to allow standalone dialplan, account and mailbox entries (STRUCTURAL) + * Added new Fields: + - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir, + - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, + - AstAccountVideoSupport, AstAccountIgnoreSDPVersion + * Removed redundant IPaddr (there's already IPAddress) + - Gives more configuration Flags for SIP-Users available (tested) + - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses + without extensibleObject (which really should be the last resort); gives + also additional possibilities for LDAP-filter + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 ------------- +------------------------------------------------------------------------------ + +Device State Handling +--------------------- + * The event infrastructure in Asterisk got another big update to help support + distributed events. It currently supports distributed device state and + distributed Voicemail MWI (Message Waiting Indication). A new module has + been merged, res_ais, which facilitates communicating events between servers. + It uses the SAForum AIS (Service Availability Forum Application Interface + Specification) CLM (Cluster Management) and EVT (Event) services to maintain + a cluster of Asterisk servers, and to share events between them. For more + information on setting this up, refer to the Distributed Device State section + of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ + +Dialplan Functions +------------------ + * Added a new dialplan function, AST_CONFIG(), which allows you to access + variables from an Asterisk configuration file. + * The JACK_HOOK function now has a c() option to supply a custom client name. + * Added two new dialplan functions from libspeex for audio gain control and + denoise, AGC() and DENOISE(). Both functions can be applied to the tx and + rx directions of a channel from the dialplan. + * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages + based on other parameters. The default is still to search based on the + forwarding station ID. However, there are new options that allow you to search + based on the message desk terminal ID, or the message desk number. + * TIMEOUT() has been modified to be accurate down to the millisecond. + * ENUM*() functions now include the following new options: + - 'u' returns the full URI and does not strip off the URI-scheme. + - 's' triggers ISN specific rewriting + - 'i' looks for branches into an Infrastructure ENUM tree + - 'd' for a direct DNS lookup without any flipping of digits. + * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa') + * CHANNEL() now has options for the maximum, minimum, and standard or normal + deviation of jitter, rtt, and loss for a call using chan_sip. + +DAHDI channel driver (chan_dahdi) Changes +---------------------------------------- + * Channels can now be configured using named sections in chan_dahdi.conf, just + like other channel drivers, including the use of templates. + * The default for pridialplan has changed from 'national' to 'unknown'. + +PBX Changes +----------- + * It is now possible to specify a pattern match as a hint. Once a phone subscribes + to something that matches the pattern a hint will be created using the contents + and variables evaluated. + * Dialplan matching has been extended to allow an extension to return to the + PBX core to wait for more digits. This is done by using the new dialplan + application called "Incomplete". This will permit a whole new level of + extension control, by giving the administrator more control over early + matches employing one of the short-circuit pattern match operators. Note + that custom applications can trigger this same behavior by returning the + special value AST_PBX_INCOMPLETE. + +Application Changes +------------------- + * Directory now permits both first and last names to be matched at the same + time. In addition, the number of digits to enter of the name can be set in + the arguments to Directory; previously, you could enter only 3, regardless + of how many names are in your company. For large companies, this should be + quite helpful. + * Voicemail now permits a mailbox setting to wrap around from first to last + messages, if the "messagewrap" option is set to a true value. + * Voicemail now permits an external script to be run, for password validation. + The script should output "VALID" or "INVALID" on stdout, depending upon the + wish to validate or invalidate the password given. Arguments are: + "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for + more details + * Dial has a new option: F(context^extension^pri), which permits a callee to + continue in the dialplan, at the specified label, if the caller hangs up. + * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the + technology name (e.g. SIP, IAX, etc) of the channel being spied on. + * The Jack application now has a c() option to supply a custom client name. + * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is + like the pre-existing whisper mode, except that the spy can also talk to the + participant on the bridged channel as well. + * Chanspy has a new option, 'n', which will allow for the spied-on party's name + to be spoken instead of the channel name or number. For more information on the + use of this option, issue the command "core show application ChanSpy" from the + Asterisk CLI. + * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between + spy modes. Use of this feature overrides the typical use of numeric DTMF. In other + words, if using the 'd' option, it is not possible to enter a number to append to + the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will + change to whisper mode, and pressing 6 will change to barge mode. + * ExternalIVR now takes several options that affect the way it performs, as + well as having several new commands. Please see the External IVR page on the Asterisk + wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB + * Added ability to communicate over a TCP socket instead of forking a child process for the + ExternalIVR application. + * ChanIsAvail has a new option, 'a', which will return all available channels instead + of just the first one if you give the function more then one channel to check. + * PrivacyManager now takes an option where you can specify a context where the + given number will be matched. This way you have more control over who is allowed + and it stops the people who blindly enter 10 digits. + * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks + answer times, disposition, on orig CDR against updates; 'D' Copies the disposition + from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the + original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes + the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(), + obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func. + * The Dial() application no longer copies the language used by the caller to the callee's + channel. If you desire for the caller's channel's language to be used for file playback + to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" . + * SendImage() no longer hangs up the channel on error; instead, it sets the + status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or + 'UNSUPPORTED'. This change makes SendImage() more consistent with other + applications. + * Park has a new option, 's', which silences the announcement of the parking space number. + * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as + invalid input and will be assumed to mean that no timeout is desired. + +SIP Changes +----------- + * Added DNS manager support to registrations for peers referencing peer entries. + DNS manager runs in the background which allows DNS lookups to be run asynchronously + as well as periodically updating the IP address. These properties allow for + better performance as well as recovery in the event of an IP change. + * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve + load/reload of large numbers of peers/users by ~40x (for large lists of peers). + These changes also provide performance improvements for call setup and tear down. + * Added ability to specify registration expiry time on a per registration basis in + the register line. + * Added support for T140 RED - redundancy in T.140 to prevent text loss due to + lost packets. + * Added t38pt_usertpsource option. See sip.conf.sample for details. + * Added SIPnotify AMI command, for sending arbitrary SIP notify commands. + * 'sip show peers' and 'sip show users' display their entries sorted in + alphabetical order, as opposed to the order they were in, in the config + file or database. + * Videosupport now supports an additional option, "always", which always sets + up video RTP ports, even on clients that don't support it. This helps with + callfiles and certain transfers to ensure that if two video phones are + connected, they will always share video feeds. + +IAX Changes +----------- + * Existing DNS manager lookups extended to check for SRV records. + * IAX2 encryption support has been improved to support periodic key rotation + within a call for enhanced security. The option "keyrotate" has been + provided to disable this functionality to preserve backwards compatibility + with older versions of IAX2 that do not support key rotation. + +CLI Changes +----------- + * New CLI command, "data get [ []]" which retrieves the + data tree based on the given . + * New CLI command "data show providers" that will display all the registered + callbacks. + * New CLI command, "config reload " which reloads any module that + references that particular configuration file. Also added "config list" + which shows which configuration files are in use. + * New CLI commands, "pri show version" and "ss7 show version" that will + display which version of libpri and libss7 are being used, respectively. + A new API call was added so trunk will now have to be compiled against + a versions of libpri and libss7 that have them or it will not know that + these libraries exist. + * The commands "core show globals", "core set global" and "core set chanvar" has + been deprecated in favor of the more semantically correct "dialplan show globals", + "dialplan set chanvar" and "dialplan set global". + * New CLI command "dialplan show chanvar" to list all variables associated + with a given channel. + +DNS manager changes +------------------- + * Addresses managed by DNS manager now can check to see if there is a DNS + SRV record for a given domain and will use that hostname/port if present. + +AMI - The manager (TCP/TLS/HTTP) +-------------------------------- + * The Status command now takes an optional list of variables to display + along with channel status. + * The QueueEntry event now also includes the channel's uniqueid + +ODBC Changes +------------ + * res_odbc no longer has a limit of 1023 total possible unshared connections, + as some people were running into this limit. This limit has been increased + to 4.2 billion. + +Queue changes +------------- + * The TRANSFER queue log entry now includes the the caller's original + position in the transferred-from queue. + * A new configuration option, "timeoutpriority" has been added. Please see the section labeled + "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option + as well as an explanation about timeout options in general + * Added a new option - C - for forcing the "answered elsewhere" flag on + cancellation of calls in to members of the queue. This is to avoid the + call to a member of a queue having the call listed as a "missed call". + +Realtime changes +---------------- + * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given + adaptive capabilities. What this means in practical terms is that if your + realtime table lacks critical fields, Asterisk will now emit warnings to + that effect. Also, some of the realtime drivers have the ability (if + configured) to automatically add those columns to the table with the + correct type and length. + +Miscellaneous +------------- + * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using + the 'setvar' option to cause a given audio file to be played upon completion + of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and + Skinny channels only. + * You can now compile Asterisk against the Hoard Memory Allocator, see the + Hoard page on the Asterisk wiki for more information: + https://wiki.asterisk.org/wiki/x/pQBB + * Config file variables may now be appended to, by using the '+=' append + operator. This is most helpful when working with long SQL queries in + func_odbc.conf, as the queries no longer need to be specified on a single + line. + * CDR config file, cdr.conf, has an added option, "initiatedseconds", + which will add a second to the billsec when the ending + time is set, if the number in the microseconds field of the end time is + greater than the number of microseconds in the answer time. This allows + users to count the 'initiated' seconds in their billing records. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 ------------- +------------------------------------------------------------------------------ + +AMI - The manager (TCP/TLS/HTTP) +-------------------------------- + * Manager has undergone a lot of changes, all of them documented + on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB + * Manager version has changed to 1.1 + * Added a new action 'CoreShowChannels' to list currently defined channels + and some information about them. + * Added a new action 'SIPshowregistry' to list SIP registrations. + * Added TLS support for the manager interface and HTTP server + * Added the URI redirect option for the built-in HTTP server + * The output of CallerID in Manager events is now more consistent. + CallerIDNum is used for number and CallerIDName for name. + * Enable https support for builtin web server. + See configs/http.conf.sample for details. + * Added a new action, GetConfigJSON, which can return the contents of an + Asterisk configuration file in JSON format. This is intended to help + improve the performance of AJAX applications using the manager interface + over HTTP. + * SIP and IAX manager events now use "ChannelType" in all cases where we + indicate channel driver. Previously, we used a mixture of "Channel" + and "ChannelDriver" headers. + * Added a "Bridge" action which allows you to bridge any two channels that + are currently active on the system. + * Added a "ListAllVoicemailUsers" action that allows you to get a list of all + the voicemail users setup. + * Added 'DBDel' and 'DBDelTree' manager commands. + * cdr_manager now reports events via the "cdr" level, separating it from + the very verbose "call" level. + * Manager users are now stored in memory. If you change the manager account + list (delete or add accounts) you need to reload manager. + * Added Masquerade manager event for when a masquerade happens between + two channels. + * Added "manager reload" command for the CLI + * Lots of commands that only provided information are now allowed under the + Reporting privilege, instead of only under Call or System. + * The IAX* commands now require either System or Reporting privilege, to + mirror the privileges of the SIP* commands. + * Added ability to retrieve list of categories in a config file. + * Added ability to retrieve the content of a particular category. + * Added ability to empty a context. + * Created new action to create a new file. + * Updated delete action to allow deletion by line number with respect to category. + * Added new action insert to add new variable to category at specified line. + * Updated action newcat to allow new category to be inserted in file above another + existing category. + * Added new event "JitterBufStats" in the IAX2 channel + * Originate now requires the Originate privilege and, if you want to call out + to a subshell, it requires the System privilege, as well. This was done to + enhance manager security. + * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" + * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details + or manager show command Atxfer from the CLI + * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more + details or manager show command IAXregistry from the CLI + +Dialplan functions +------------------ + * Added the DEVICE_STATE() dialplan function which allows retrieving any device + state in the dialplan, as well as creating custom device states that are + controllable from the dialplan. + * Extend CALLERID() function with "pres" and "ton" parameters to + fetch string representation of calling number presentation indicator + and numeric representation of type of calling number value. + * MailboxExists converted to dialplan function + * A new option to Dial() for telling IP phones not to count the call + as "missed" when dial times out and cancels. + * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan + mutex. No deadlocks are possible, as LOCK() only allows a single lock to be + held for any given channel. Also, locks are automatically freed when a + channel is hung up. + * Added HINT() dialplan function that allows retrieving hint information. + Hints are mappings between extensions and devices for the sake of + determining the state of an extension. This function can retrieve the list + of devices or the name associated with a hint. + * Added EXTENSION_STATE() dialplan function which allows retrieving the state + of any extension. + * Added SYSINFO() dialplan function which allows retrieval of system information + * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for + the existence of a dialplan target. + * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to + upper and lower case, respectively. + * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique + ID for the call (not the Asterisk call ID or unique ID), provided that the + channel driver supports this. For SIP, you get the SIP call-ID for the + bridged channel which you can store in the CDR with a custom field. + +CLI Changes +----------- + * Added CLI permissions, config file: cli_permissions.conf + default is to allow all commands for every local user/group. + Also this new feature added three new CLI commands: + - cli check permissions {|@|@} [] + - cli reload permissions + - cli show permissions + * New CLI command "core show hint" (usage: core show hint ) + * New CLI command "core show settings" + * Added 'core show channels count' CLI command. + * Added the ability to set the core debug and verbose values on a per-file basis. + * Added 'queue pause member' and 'queue unpause member' CLI commands + * Ability to set process limits ("ulimit") without restarting Asterisk + * Enhanced "agi debug" to print the channel name as a prefix to the debug + output to make debugging on busy systems much easier. + * New CLI commands "dialplan set extenpatternmatching true/false" + * New CLI command: "core set chanvar" to set a channel variable from the CLI. + * Added an easy way to execute Asterisk CLI commands at startup. Any commands + listed in the startup_commands section of cli.conf will get executed. + * Added a CLI command, "devstate change", which allows you to set custom device + states from the func_devstate module that provides the DEVICE_STATE() function + and handling of the "Custom:" devices. + * New CLI command: "sip show sched" which shows all ast_sched entries for sip, + sorted into the different possible callbacks, with the number of entries + currently scheduled for each. Gives you a feel for how busy the sip channel + driver is. + * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel. + * Cleanup another bunch of CLI commands. Now all modules follow the same schema. + (Done by lmadsen, junky and mvanbaak during the devcon 2008) + +SIP changes +----------- + * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this + option is enabled, Asterisk will watch for a CNG tone in the incoming audio + for a received call. If it is detected, the channel will jump to the + 'fax' extension in the dialplan. + * The default SIP useragent= identifier now includes the Asterisk version + * A new option, match_auth_username in sip.conf changes the matching of incoming requests. + If set, and the incoming request carries authentication info, + the username to match in the users list is taken from the Digest header + rather than from the From: field. This feature is considered experimental. + * The "musiconhold" and "musicclass" settings in sip.conf are now removed, + since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4 + * The "localmask" setting was removed in version 1.2 and the reminder about it + being removed is now also removed. + * A new option "busylevel" for setting a level of calls where asterisk reports + a device as busy, to separate it from call-limit. This value is also added + to the SIP_PEER dialplan function. + * A new realtime family called "sipregs" is now supported to store SIP registration + data. If this family is defined, "sippeers" will be used for configuration and + "sipregs" for registrations. If it's not defined, "sippeers" will be used for + registration data, as before. + * The SIPPEER function have new options for port address, call and pickup groups + * Added support for T.140 realtime text in SIP/RTP + * The "checkmwi" option has been removed from sip.conf, as it is no longer + required due to the restructuring of how MWI is handled. See the descriptions + in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf + for more information. + * Added rtpdest option to CHANNEL() dialplan function. + * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place. + * SIP now adds a header to the CANCEL if the call was answered by another phone + in the same dial command, or if the new c option in dial() is used. + * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically + states it is not needed. For phones, however, that do require it the "registertrying" option + has been added so it can be enabled. + * A new option called "callcounter" (global/peer/user level) enables call counters needed + for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously + used to enable this functionality). + * New settings for timer T1 and timer B on a global level or per device. This makes it + possible to force timeout faster on non-responsive SIP servers. These settings are + considered advanced, so don't use them unless you have a problem. + * Added a dial string option to be able to set the To: header in an INVITE to any + SIP uri. + * Added a new global and per-peer option, qualifyfreq, which allows you to configure + the qualify frequency. + * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that + were not properly torn down due to network or endpoint failures during an established + SIP session. + * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB + and configs/sip.conf.sample for more information on how it is used. + * Added a new configuration option "authfailureevents" that enables manager events when + a peer can't authenticate properly. + * Added DNS manager support to registrations for peers not referencing a peer entry. + +IAX2 changes +------------ + * Added the trunkmaxsize configuration option to chan_iax2. + * Added the srvlookup option to iax.conf + * Added support for OSP. The token is set and retrieved through the CHANNEL() + dialplan function. + +XMPP Google Talk/Jingle changes +------------------------------- + * Added the bindaddr option to gtalk.conf. + +Skinny changes +------------- + * Added skinny show device, skinny show line, and skinny show settings CLI commands. + * Proper codec support in chan_skinny. + * Added settings for IP and Ethernet QoS requests + +MGCP changes +------------ + * Added separate settings for media QoS in mgcp.conf + +Console Channel Driver changes +------------------------------ + * Added experimental support for video send & receive to chan_oss. + This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as + a video source. + +Phone channel changes (chan_phone) +---------------------------------- + * Added G729 passthrough support to chan_phone for Sigma Designs boards. + +H.323 channel Changes +--------------------- + * H323 remote hold notification support added (by NOTIFY message + and/or H.450 supplementary service) + +Local channel changes +--------------------- + * The device state functionality in the Local channel driver has been updated + to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed + to just UNKNOWN if the extension exists. + * Added jitterbuffer support for chan_local. This allows you to use the + generic jitterbuffer on incoming calls going to Asterisk applications. + For example, this would allow you to use a jitterbuffer for an incoming + SIP call to Voicemail by putting a Local channel in the middle. This + feature is enabled by using the 'j' option in the Dial string to the Local + channel in conjunction with the existing 'n' option for local channels. + * A 'b' option has been added which causes chan_local to return the actual channel + that is behind it when queried. This is useful for transfer scenarios as the + actual channel will be transferred, not the Local channel. + +Agent channel changes +---------------------- + * The ackcall and endcall options are now supplemented with options acceptdtmf + and enddtmf. These allow for the DTMF keypress to be configurable. The options + default to their old hard-coded values ('#' and '*' respectively) so this should + not break any existing agent installations. + +DAHDI channel driver (chan_dahdi) Changes +---------------------------------------- + * SS7 support (via libss7 library) + * In India, some carriers transmit CID via dtmf. Some code has been added + that will handle some situations. The cidstart=polarity_IN choice has been added for + those carriers that transmit CID via dtmf after a polarity change. + * CID matching information is now shown when doing 'dialplan show'. + * Added dahdi show version CLI command. + * Added setvar support to chan_dahdi.conf channel entries. + * Added two new options: mwimonitor and mwimonitornotify. These options allow + you to enable MWI monitoring on FXO lines. When the MWI state changes, + the script specified in the mwimonitornotify option is executed. An internal + event indicating the new state of the mailbox is also generated, so that + the normal MWI facilities in Asterisk work as usual. + * Added signalling type 'auto', which attempts to use the same signalling type + for a channel as configured in DAHDI. This is primarily designed for analog + ports, but will also work for digital ports that are configured for FXS or FXO + signalling types. This mode is also the default now, so if your chan_dahdi.conf + does not specify signalling for a channel (which is unlikely as the sample + configuration file has always recommended specifying it for every channel) then + the 'auto' mode will be used for that channel if possible. + * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb + state for a channel; also ensured that the DNDState Manager event is + emitted no matter how the DND state is set or cleared. + +New Channel Drivers +------------------- + * Added a new channel driver, chan_unistim. See the Asterisk wiki at + https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample + for details. This new channel driver allows you to use Nortel i2002, + i2004, and i2050 phones with Asterisk. + * Added a new channel driver, chan_console, which uses portaudio as a cross + platform audio interface. It was written as a channel driver that would + work with Mac CoreAudio, but portaudio supports a number of other audio + interfaces, as well. Note that this channel driver requires v19 or higher + of portaudio; older versions have a different API. + +DUNDi changes +------------- + * Added the ability to specify arguments to the Dial application when using + the DUNDi switch in the dialplan. + * Added the ability to set weights for responses dynamically. This can be + done using a global variable or a dialplan function. Using the SHELL() + function would allow you to have an external script set the weight for + each response. + * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These + functions will allow you to initiate a DUNDi query from the dialplan, + find out how many results there are, and access each one. + * Added the ability to specify a port for a dundi peer. + +ENUM changes +------------ + * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These + functions will allow you to initiate an ENUM lookup from the dialplan, + and Asterisk will cache the results. ENUMRESULT can be used to access + the results without doing multiple DNS queries. + +Voicemail Changes +----------------- + * Added the ability to customize which sound files are used for some of the + prompts within the Voicemail application by changing them in voicemail.conf + * Added the ability for the "voicemail show users" CLI command to show users + configured by the dynamic realtime configuration method. + * MWI (Message Waiting Indication) handling has been significantly + restructured internally to Asterisk. It is now totally event based + instead of polling based. The voicemail application will notify other + modules that have subscribed to MWI events when something in the mailbox + changes. + This also means that if any other entity outside of Asterisk is changing + the contents of mailboxes, then the voicemail application still needs to + poll for changes. Examples of situations that would require this option + are web interfaces to voicemail or an email client in the case of using + IMAP storage. So, two new options have been added to voicemail.conf + to account for this: "pollmailboxes" and "pollfreq". See the sample + configuration file for details. + * Added "tw" language support + * Added support for storage of greetings using an IMAP server + * Added ability to customize forward, reverse, stop, and pause keys for message playback + * SMDI is now enabled in voicemail using the smdienable option. + * A "lockmode" option has been added to asterisk.conf to configure the file + locking method used for voicemail, and potentially other things in the + future. The default is the old behavior, lockfile. However, there is a + new method, "flock", that uses a different method for situations where the + lockfile will not work, such as on SMB/CIFS mounts. + * Added the ability to backup deleted messages, to ease recovery in the case + that a user accidentally deletes a message, and discovers that they need it. + * Reworked the SMDI interface in Asterisk. The new way to access SMDI information + is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file + smdi.conf can now be configured with options to map SMDI station IDs to Asterisk + voicemail boxes. The SMDI interface can also poll for MWI changes when some + outside entity is modifying the state of the mailbox (such as IMAP storage or + a web interface of some kind). + * Added the support for marking messages as "urgent." There are two methods to accomplish + this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent + is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark + the message as urgent after he has recorded a voicemail by following the voice instructions. + When listening to voicemails using VoiceMailMain urgent messages will be presented before other + messages + * Added "is" language support + +Queue changes +------------- + * Added the general option 'shared_lastcall' so that member's wrapuptime may be + used across multiple queues. + * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and + setqueueentryvar options for each queue, see queues.conf.sample for details. + * Added keepstats option to queues.conf which will keep queue + statistics during a reload. + * setinterfacevar option in queues.conf also now sets a variable + called MEMBERNAME which contains the member's name. + * Added 'Strategy' field to manager event QueueParams which represents + the queue strategy in use. + * Added option to run macro when a queue member is connected to a caller, + see queues.conf.sample for details. + * app_queue now has a 'loose' option which is almost exactly like 'strict' except it + does not count paused queue members as unavailable. + * Added min-announce-frequency option to queues.conf which allows you to control the + minimum amount of time between queue announcements for use when the caller's queue + position changes frequently. + * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the + queue log. + * Added ability for non-realtime queues to have realtime members + * Added the "linear" strategy to queues. + * Added the "wrandom" strategy to queues. + * Added new channel variable QUEUE_MIN_PENALTY + * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining + rules in queuerules.conf. See configs/queuerules.conf.sample for details + * Added a new parameter for member definition, called state_interface. This may be + used so that a member may be called via one interface but have a different interface's + device state reported. + * Added new CLI and Manager commands relating to reloading queues. From the CLI, see + "queue reload", "queue reset stats". Also see "manager show command QueueReload" and + "manager show command QueueReset." + * New configuration option: randomperiodicannounce. If a list of periodic announcements is + specified by the periodic-announce option, then one will be chosen randomly when it is time + to play a periodic announcment + * New configuration options: announce-position now takes two more values in addition to "yes" and + "no." Two new options, "limit" and "more," are allowed. These are tied to another option, + announce-position-limit. By setting announce-position to "limit" callers will only have their + position announced if their position is less than what is specified by announce-position-limit. + If announce-position is set to "more" then callers beyond the position specified by announce-position-limit + will be told that their are more than announce-position-limit callers waiting. + * Two new queue log events have been added. An ADDMEMBER event will be logged + when a realtime queue member is added and a REMOVEMEMBER event will be logged + when a realtime queue member is removed. Since there is no calling channel associated + with these events, the string "REALTIME" is placed where the channel's unique id + is typically placed. + * The configuration method for the "joinempty" and "leavewhenempty" options has + changed to a comma-separated list of methods of determining member availability + instead of vague terms such as "yes," "loose," "no," and "strict." These old four + values are still accepted for backwards-compatibility, though. + * The average talktime is now calculated on queues. This information is reported via the + CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary, + and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for + the queue. + +MeetMe Changes +-------------- + * The 'o' option to provide an optimization has been removed and its functionality + has been enabled by default. + * When a conference is created, the UNIQUEID of the channel that caused it to be + created is stored. Then, every channel that joins the conference will have the + MEETMEUNIQUEID channel variable set with this ID. This can be used to relate + callers that come and go from long standing conferences. + * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin, + except it does operations on a channel by name, instead of number in a conference. + This is a very useful feature in combination with the 'X' option to ChanSpy. + * Added 'C' option to Meetme which causes a caller to continue in the dialplan + when kicked out. + * Added new RealTime functionality to provide support for scheduled conferencing. + This includes optional messages to the caller if they attempt to join before + the schedule start time, or to allow the caller to join the conference early. + Also included is optional support for limiting the number of callers per + RealTime conference. + * Added the S() and L() options to the MeetMe application. These are pretty + much identical to the S() and L() options to Dial(). They let you set + timeouts for the conference, as well as have warning sounds played to + let the caller know how much time is left, and when it is running out. + * Added the ability to do "meetme concise" with the "meetme" CLI command. + This extends the concise capabilities of this CLI command to include + listing all conferences, instead of an addition to the other sub commands + for the "meetme" command. + * Added the ability to specify the music on hold class used to play into the + conference when there is only one member and the M option is used. + * Added MEETME_INFO dialplan function which provides a way to query + various properties of a Meetme conference. + * Added new admin features: *81: Roll call, *82: eject all, *83: mute all, + and *84: record in-conf + +Other Dialplan Application Changes +---------------------------------- + * Argument support for Gosub application + * From the to-do lists: straighten out the app timeout args: + Wait() app now really does 0.3 seconds- was truncating arg to an int. + WaitExten() same as Wait(). + Congestion() - Now takes floating pt. argument. + Busy() - now takes floating pt. argument. + Read() - timeout now can be floating pt. + WaitForRing() now takes floating pt timeout arg. + SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds. + * Added 's' option to Page application. + * Added an optional timeout argument to the Page application. + * Added 'E', 'V', and 'P' commands to ExternalIVR. + * Added 'o' and 'X' options to Chanspy. + * Added a new dialplan application, Bridge, which allows you to bridge the + calling channel to any other active channel on the system. + * Added the ability to specify a music on hold class to play instead of ringing + for the SLATrunk application. + * The Read application no longer exits the dialplan on error. Instead, it sets + READSTATUS to ERROR, which you can catch and handle separately. + * Added 'm' option to Directory, which lists out names, 8 at a time, instead + of asking for verification of each name, one at a time. + * Privacy() no longer uses privacy.conf, as all options are specifiable as + direct options to the app. + * AMD() has a new "maximum word length" option. "show application AMD" from the CLI + for more details + * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications + * The ChannelRedirect application no longer exits the dialplan if the given channel + does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success + or NOCHANNEL if the given channel was not found. + * The silencethreshold setting that was previously configurable in multiple + applications is now settable globally via dsp.conf. + +Music On Hold Changes +--------------------- + * A new option, "digit", has been added for music on hold classes in + musiconhold.conf. If this is set for a music on hold class, a caller + listening to music on hold can press this digit to switch to listening + to this music on hold class. + * Support for realtime music on hold has been added. + * In conjunction with the realtime music on hold, a general section has + been added to musiconhold.conf, its sole variable is cachertclasses. If this + is set, then music on hold classes found in realtime will be cached in memory. + +AEL Changes +----------- + * AEL upgraded to use the Gosub with Arguments instead + of Macro application, to hopefully reduce the problems + seen with the artificially low stack ceiling that + Macro bumps into. Macros can only call other Macros + to a depth of 7. Tests run using gosub, show depths + limited only by virtual memory. A small test demonstrated + recursive call depths of 100,000 without problems. + -- in addition to this, all apps that allowed a macro + to be called, as in Dial, queues, etc, are now allowing + a gosub call in similar fashion. + * AEL now generates LOCAL(argname) declarations when it + Set()'s the each arg name to the value of ${ARG1}, ${ARG2), + etc. That makes the arguments local in scope. The user + can define their own local variables in macros, now, + by saying "local myvar=someval;" or using Set() in this + fashion: Set(LOCAL(myvar)=someval); ("local" is now + an AEL keyword). + * utils/conf2ael introduced. Will convert an extensions.conf + file into extensions.ael. Very crude and unfinished, but + will be improved as time goes by. Should be useful for a + first pass at conversion. + * aelparse will now read extensions.conf to see if a referenced + macro or context is there before issuing a warning. + * AEL parser sets a local channel variable ~~EXTEN~~, to + preserve the value of ${EXTEN} thru switch statements. + * New operator in $[...] expressions: the ~~ operator serves + as a concatenation operator. AT THE MOMENT, it is really only + necessary and useful in AEL, especially in if() expressions. + Operation: ${a} ~~ ${b| with force both a and b to strings, strip + any enclosing double-quotes, and evaluate to the value of a + concatenated with the value of b. For example if a is set to + "xyz" and b has the value "abc", then ${a} ~~ ${b| would + evaluate to xyzabc . + + +Call Features (res_features) Changes +------------------------------------ + * Added the parkedcalltransfers option to features.conf + * Added parkedcallparking option to control one touch parking w/ parking + pickup + * Added parkedcallhangup option to control disconnect feature w/ parking + pickup + * Added parkedcallrecording option to control one-touch record w/ parking + pickup + * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and + parkedcalltransfers option support for multiple parking lots. + * Added BRIDGE_FEATURES variable to set available features for a channel + * The built-in method for doing attended transfers has been updated to + include some new options that allow you to have the transferee sent + back to the person that did the transfer if the transfer is not successful. + See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries" + in features.conf.sample. + * Added support for configuring named groups of custom call features in + features.conf. This means that features can be written a single time, and + then mapped into groups of features for different key mappings or easier + access control. + * Updated the ParkedCall application to allow you to not specify a parking + extension. If you don't specify a parking space to pick up, it will grab + the first one available. + * Added cli command 'features reload' to reload call features from features.conf + * Moved into core asterisk binary. + * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms. + * Added the ability for custom parking lots to be configured with their own + parking extension with the parkext option. + +Language Support Changes +------------------------ + * Brazilian Portuguese (pt-BR) in VM, and say.c was added + * Added support for the Hungarian language for saying numbers, dates, and times. + +AGI Changes +----------- + * Added SPEECH commands for speech recognition. A complete listing can be found + using agi show. + * If app_stack is loaded, GOSUB is a native AGI command that may be used to + invoke subroutines in the dialplan. Note that calling EXEC with Gosub + does not behave as expected; the native command needs to be used, instead. + * Added the ability to perform SRV lookups on fast AGI calls. To use this + feature, simply use hagi: instead of agi: as the protocol portion + of the URI parameter to the AGI function call in your dial plan. Also note + that specifying a port number in the AGI URI will disable SRV lookups, + even if you use the hagi: protocol. + * No longer support MSG_OOB flag on HANGUP. + +Logger changes +-------------- + * Added rotatestrategy option to logger.conf, along with two new options: + "timestamp" which will use the time to name the logger files instead of + sequence number; and "rotate", which rotates the names of the log files, + similar to the way syslog rotates files. + * Added exec_after_rotate option to logger.conf, which allows a system + command to be run after rotation. This is primarily useful with + rotatestrategy=rotate, to allow a limit on the number of log files kept + and to ensure that the oldest log file gets deleted. + * Added realtime support for the queue log + +Call Detail Records +------------------- + * The cdr_manager module has a [mappings] feature, like cdr_custom, + to add fields to the manager event from the CDR variables. + * Added cdr_adaptive_odbc, a new module that adapts to the structure of your + backend database CDR table. Specifically, additional, non-standard + columns are supported, merely by setting the corresponding CDR variable in + your dialplan. In addition, you may alias any column to another name (for + example, if you want the 'src' CDR variable to be column 'ANI' in the DB, + simply "alias src => ANI" in the configuration file). Records may be + posted to more than one backend, simply by specifying multiple categories + in the configuration file. And finally, you may filter which CDRs get + posted to each backend, by specifying a filter (which the record must + match) for the particular category. Filters are additive (meaning all + rules must match to post that CDR). + * The Postgres CDR module now supports some features of the cdr_adaptive_odbc + module. Specifically, you may add additional columns into the table and + they will be set, if you set the corresponding CDR variable name. Also, + if you omit columns in your database table, they will be silently skipped + (but a record will still be inserted, based on what columns remain). Note + that the other two features from cdr_adaptive_odbc (alias and filter) are + not currently supported. + * The ResetCDR application now has an 'e' option that re-enables a CDR if it + has been disabled using the NoCDR application. + +Miscellaneous New Modules +------------------------- + * Added a new CDR module, cdr_sqlite3_custom. + * Added a new realtime configuration module, res_config_sqlite + * Added a new codec translation module, codec_resample, which re-samples + signed linear audio between 8 kHz and 16 kHz to help support wideband + codecs. + * Added a new module, res_phoneprov, which allows auto-provisioning of phones + based on configuration templates that use Asterisk dialplan function and + variable substitution. It should be possible to create phone profiles and + templates that work for the majority of phones provisioned over http. It + is currently only intended to provision a single user account per phone. + An example profile and set of templates for Polycom phones is provided. + NOTE: Polycom firmware is not included, but should be placed in + AST_DATA_DIR/phoneprov/configs to match up with the included templates. + * Added a new module, app_jack, which provides interfaces to JACK, the Jack + Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are + provided; there is a JACK() application, and a JACK_HOOK() function. Both + interfaces create an input and output JACK port. The application makes + these ports the endpoint of the call. The audio coming from the channel + goes out the output port and whatever comes back in on the input port is + what gets sent to the channel. The JACK_HOOK() function turns on a JACK + audiohook on the channel. This lets you run the audio coming from a + channel through JACK, and whatever comes back in is what gets forwarded + on as the channel's audio. This is very useful for building custom + vocoders or doing recording or analysis of the channel's audio in another + application. + * Added a new module, res_config_curl, which permits using a HTTP POST url + to retrieve, create, update, and delete realtime information from a remote + web server. Note that this module requires func_curl.so to be loaded for + backend functionality. + * Added a new module, res_config_ldap, which permits the use of an LDAP + server for realtime data access. + * Added support for writing and running your dialplan in lua using the pbx_lua + module. See configs/extensions.lua.sample for examples of how to do this. + +Miscellaneous +------------- + * Ability to use libcap to set high ToS bits when non-root + on Linux. If configure is unable to find libcap then you + can use --with-cap to specify the path. + * Added maxfiles option to options section of asterisk.conf which allows you to specify + what Asterisk should set as the maximum number of open files when it loads. + * Added the jittertargetextra configuration option. + * Added support for setting the CoS for VLAN traffic (802.1p). See the sample + configuration files for the IP channel drivers. The new option is "cos". + This information is also documented on the Asterisk wiki at + https://wiki.asterisk.org/wiki/x/EYBG + * When originating a call using AMI or pbx_spool that fails the reason for failure + will now be available in the failed extension using the REASON dialplan variable. + * Added support for reading the TOUCH_MONITOR_PREFIX channel variable. + It allows you to configure a prefix for auto-monitor recordings. + * A new extension pattern matching algorithm, based on a trie, is introduced + here, that could noticeably speed up mid-sized to large dialplans. + It is NOT used by default, as duplicating the behaviour of the old pattern + matcher is still under development. A config file option, in extensions.conf, + in the [general] section, called "extenpatternmatchingnew", is by default + set to false; setting that to true will force the use of the new algorithm. + Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can + be used to switch the algorithms at run time. + * A new option when starting a remote asterisk (rasterisk, asterisk -r) for + specifying which socket to use to connect to the running Asterisk daemon + (-s) + * Performance enhancements to the sched facility, which is used in + the channel drivers, etc. Added hashtabs and doubly-linked lists + to speed up deletion; start at the beginning or end of list to + speed up insertion. + * Added Doubly-linked lists after the fashion of linkedlists.h. They are in + dlinkedlists.h. Doubly-linked lists feature fast deletion times. + Added regression tests to the tests/ dir, also. + * Added a refcount trace feature to astobj2 for those trying to balance + object creation, deletion; work, play; space and time. See the + notes in astobj2.h. Also, see utils/refcounter as well, as a + quick way to find unbalanced refcounts in what could be a sea + of objects that were balanced. + * Added logging to 'make update' command. See update.log + * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that + do not come from the remote party. + * Added the 'n' option to the SpeechBackground application to tell it to not + answer the channel if it has not already been answered. + * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be + turned on, via the CHANNEL(trace) dialplan function. Could be useful for + dialplan debugging. + * iLBC source code no longer included (see UPGRADE.txt for details) + * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if + deadlock is detected, a backtrace of the stack which led to the lock calls + will be output to the CLI. + * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing + the "core show locks" CLI command will give lock information output as well + as a backtrace of the stack which led to the lock calls. + * users.conf now sports an optional alternateexts property, which permits + allocation of additional extensions which will reach the specified user. + * A new option for the configure script, --enable-internal-poll, has been added + for use with systems which may have a buggy implementation of the poll system + call. If you notice odd behavior such as the CLI being unresponsive on remote + consoles, you may want to try using this option. This option is enabled by default + on Darwin systems since it is known that the Darwin poll() implementation has + odd issues. + +Timer Changes +-------------------- +* In addition to timing from DAHDI, there is a new timing module called + res_timing_timerfd. In order to use this, you must be running Linux with + a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure + script will be able to tell if you have the requirements. From menuselect, select + res_timing_timerfd from the Resource Modules menu. diff --git a/UPGRADE.txt b/UPGRADE.txt index f453d9e296..972e4fb17a 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -1,6 +1,3 @@ -===== WARNING, THIS FILE IS OBSOLETE AND WILL BE REMOVED IN A FUTURE VERSION ===== -See 'Upgrade Notes' in the CHANGES file - =========================================================== === === THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE