This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
4234 lines
211 KiB
Text
4234 lines
211 KiB
Text
==============================================================================
|
||
===
|
||
=== 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 12 to Asterisk 13 --------------------
|
||
------------------------------------------------------------------------------
|
||
|
||
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.
|
||
|
||
------------------------------------------------------------------------------
|
||
--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
|
||
------------------------------------------------------------------------------
|
||
|
||
AgentRequest
|
||
------------------
|
||
* Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
|
||
connect with an incoming caller after being alerted to the presence
|
||
of the incoming caller. The most likely reason this would happen is
|
||
the agent did not acknowledge the call in time.
|
||
|
||
AMI
|
||
------------------
|
||
* New events have been added for the TALK_DETECT function. When the function
|
||
is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
|
||
emitted to connected AMI clients indicating the start/stop of talking on
|
||
the channel.
|
||
|
||
ARI
|
||
------------------
|
||
* New event models have been aded for the TALK_DETECT function. When the
|
||
function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
|
||
events will be emitted to connected WebSockets subscribed to the channel,
|
||
indicating the start/stop of talking on the channel.
|
||
|
||
Functions
|
||
------------------
|
||
* A new function, TALK_DETECT, has been added. When set on a channel, this
|
||
fucntion causes events indicating the starting/stoping of talking on said
|
||
channel to be emitted to both AMI and ARI clients.
|
||
|
||
------------------------------------------------------------------------------
|
||
--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
|
||
------------------------------------------------------------------------------
|
||
|
||
ARI
|
||
------------------
|
||
* A new Playback URI 'tone' has been added. Tones are specified either as
|
||
an indication name (e.g. 'tone:busy') from indications.conf or as a tone
|
||
pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
|
||
URIs in that they must be stopped manually and will continue to occupy
|
||
a channel's ARI control queue until they are stopped. They also can not
|
||
be rewound or fastforwarded.
|
||
|
||
* User events can now be generated from ARI. Events can be signalled with
|
||
arbitrary json variables, and include one or more of channel, bridge, or
|
||
endpoint snapshots. An application must be specified which will receive
|
||
the event message (other applications can subscribe to it). The message
|
||
will also be delivered via AMI provided a channel is attached. Dialplan
|
||
generated user event messages are still transmitted via the channel, and
|
||
will only be received by a stasis application they are attached to or if
|
||
the channel is subscribed to.
|
||
|
||
chan_sip
|
||
-----------
|
||
* SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
|
||
fields for prohibited callingpres information. Values are legacy, no, and
|
||
yes. By default, legacy is used.
|
||
trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
|
||
dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
|
||
headers are appended to outbound SIP messages just as they are with
|
||
allowed callingpres values, but data about the remote party's identity is
|
||
anonymized.
|
||
When sendrpid=rpid, only the remote party's domain is anonymized.
|
||
trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
|
||
headers are not sent.
|
||
trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
|
||
party information in tact even for prohibited callingpres information.
|
||
In the case of PAI, a Privacy: id header will be appended for prohibited
|
||
calling information to communicate that the private information should
|
||
not be relayed to untrusted parties.
|
||
|
||
res_parking
|
||
------------------
|
||
* Manager action 'Park' now takes an additional argument 'AnnounceChannel'
|
||
which can be used to announce the parked call's location to an arbitrary
|
||
channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
|
||
parties in a one to one bridge, 'TimeoutChannel' is treated as having
|
||
parked 'Channel' like with the Park Call DTMF feature and will receive
|
||
announcements prior to being hung up.
|
||
|
||
------------------------------------------------------------------------------
|
||
--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
|
||
------------------------------------------------------------------------------
|
||
|
||
Applications
|
||
--------------------------
|
||
* 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'
|
||
* Monitor() - A new option, B(), has been added that will turn on a periodic
|
||
beep while the call is being recorded.
|
||
|
||
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.
|
||
|
||
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.
|
||
|
||
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.
|
||
|
||
Debugging
|
||
-------------------------
|
||
* 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.
|
||
|
||
Core
|
||
------------------
|
||
* Exposed sorcery-based configuration files like pjsip.conf to dialplans via
|
||
the new AST_SORCERY diaplan function.
|
||
|
||
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.
|
||
|
||
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 <conference> 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, "<ZOMBIE>" 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:<agent-id> 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 <parkinglot>' will also show the parked calls in a specific
|
||
parking lot.
|
||
|
||
* The CLI command 'parkedcalls' is now deprecated in favor of
|
||
'parking show <parkinglot>'.
|
||
|
||
* 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/<client name>/<buddy address>
|
||
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
|
||
<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.
|
||
|
||
* 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<75>e"
|
||
|
||
* 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 ----------------
|
||
------------------------------------------------------------------------------
|
||
|
||
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,<slave-channel-name>) 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=<cause code> 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:<name>' to
|
||
retrieve state for a particular bridge, where <name> 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" <number> 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(<keypad_digits>)
|
||
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(<keypad_digits>)
|
||
dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
|
||
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>/<destination>/<control address>
|
||
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 <path> [<search> [<filter>]]" which retrieves the
|
||
data tree based on the given <path>.
|
||
* New CLI command "data show providers" that will display all the registered
|
||
callbacks.
|
||
* New CLI command, "config reload <file.conf>" 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 {<username>|@<groupname>|<username>@<groupname>} [<command>]
|
||
- cli reload permissions
|
||
- cli show permissions
|
||
* New CLI command "core show hint" (usage: core show hint <exten>)
|
||
* 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
|
||
|
||
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.
|