In Brief

Asterisk 11 introduces several new features that build on the features in Asterisk 10. Highlights include:

• Call Identifier Logging, which makes it easier for system admins to debug problems in their deployments
• Named Callgroups and Pickupgroups, which effectively removes the 64 callgroup/pickupgroup limitation
• General and Technology Specific Hangup Cause querying from the dialplan
• Hangup Handlers, which simplify hangup routine implementations in the dialplan
• Pre-dial routines, which let a dialplan writer execute a routine on a channel during construction
• Initial support for WebRTC, including support for ICE, DTLS-SRTP, and SIP over Websockets
• Named ACLs, which let ACLs be defined once and referred to by name from any module
• A brand new XMPP channel driver chan_motif, which has revamped support for the various Google Jingle protocols

And much more. A full listing is below.

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" or 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 call to Bridge 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.

Dial

• Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller channels respectively before the callee channels are called. See pre-dial handlers for more information.

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 FollowMe. By setting this option, Asterisk will not update the caller with connected line changes when they occur. This is similar to options in Dial and 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. For more information, see pre-dial handlers.
• 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 option in Dial, 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.

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 <astspooldir>/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 or Upgrading to Asterisk 11 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.
• MESSAGE(from) for incoming SIP messages now returns "display-name" <uri> 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.

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.

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.
• 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.
• Add 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.

chan_skinny

• Added skinny version 17 protocol support.

chan_unistim

• 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.

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 this option was set to a hard coded value in the 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.

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 can now 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. See manager.conf.sample.

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 <channel> 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 command 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 detailed 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 <entity> <state>[,<subtype>[,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, e.g., files, mp3, etc.

Parking

• New per parking lot options: comebackcontext and comebackdialtime. See 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 processing 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 enabled, 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 using the icesupport setting; it is disabled by default. 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 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 Upgrading to Asterisk 11 or 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.