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==============================================================================
===
=== 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 11 to Asterisk 12 --------------------
------------------------------------------------------------------------------
AMI (Asterisk Manager Interface)
------------------
* 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.
* 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.
Features
-------------------
* 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 automixmonitor to the BRIDGE_FEATURES channel variable.
Logging
-------------------
* 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.
Queue
-------------------
* Add queue available hint. exten => 8501,hint,Queue:markq_avail
Note: the suffix '_avail' after the queuename.
Reports 'InUse' for no logged in agents or no free agents.
Reports 'Idle' when an agent is free.
Core
------------------
* 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.
------------------------------------------------------------------------------
--- 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.
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.
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 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" <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.
* 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.
* 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.
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.
* 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 <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" 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 <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 (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 ----------------
------------------------------------------------------------------------------