============================================================================== === === 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 16 to Asterisk 17 -------------------- ------------------------------------------------------------------------------ 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 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. 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. ------------------------------------------------------------------------------ --- 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 unlikey) 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 negotation 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 annouced 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( 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 ''. 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 ''. Where is something like ''. 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 ( but rtpengine to eth1 ( * 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( 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 altert 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@@ 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 inhereted 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 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, CONFBRIGE_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 canflicts 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. Functionalty 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 exection 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 Briding 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 offerred 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 MixMontior 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 enginer (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 functionaility 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 sytems, 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 supporing 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 semanticly 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 specifiy 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 specifyable 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 issueing 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.