Review: https://reviewboard.asterisk.org/r/1075/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1722 lines
98 KiB
Text
1722 lines
98 KiB
Text
==============================================================================
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===
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=== This file documents the new and/or enhanced functionality added in
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=== the Asterisk versions listed below. This file does NOT include
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=== changes in behavior that would not be backwards compatible with
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=== previous versions; for that information see the UPGRADE.txt file
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=== and the other UPGRADE files for older releases.
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
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------------------------------------------------------------------------------
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Asterisk Manager Interface
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--------------------------
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* PeerStatus now includes Address and Port.
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* Added Hold events for when the remote party puts the call on and off hold
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for chan_dahdi ISDN channels.
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Asterisk HTTP Server
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--------------------------
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* The HTTP Server can bind to IPv6 addresses.
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CLI Changes
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--------------------------
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* New 'gtalk show settings' command showing the current settings loaded from
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gtalk.conf.
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* The 'logger reload' command now supports an optional argument, specifying an
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alternate configuration file to use.
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CDR
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--------------------------
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* The filter option in cdr_adaptive_odbc now supports negating the argument,
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thus allowing records which do NOT match the specified filter.
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Dialplan Variables
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------------------
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* Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
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ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
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variables from asterisk.conf.
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libpri channel driver (chan_dahdi) DAHDI changes
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--------------------------
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* Added moh_signaling option to specify what to do when the channel's bridged
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peer puts the ISDN channel on hold.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
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------------------------------------------------------------------------------
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SIP Changes
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-----------
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* Added preferred_codec_only option in sip.conf. This feature limits the joint
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codecs sent in response to an INVITE to the single most preferred codec.
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* Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
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to be used for the outgoing call. It must be one of the codecs configured
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for the device.
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* Added tlsprivatekey option to sip.conf. This allows a separate .pem file
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to be used for holding a private key. If tlsprivatekey is not specified,
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tlscertfile is searched for both public and private key.
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* Added tlsclientmethod option to sip.conf. This allows the protocol for
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outbound client connections to be specified.
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* The sendrpid parameter has been expanded to include the options
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'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
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header to be sent (equivalent to setting sendrpid=yes) and setting
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sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
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* The 'ignoresdpversion' behavior has been made automatic when the SDP received
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is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
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since the call will fail if Asterisk does not process the incoming SDP, Asterisk
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will accept the SDP even if the SDP version number is not properly incremented,
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but will generate a warning in the log indicating that the SIP peer that sent
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the SDP should have the 'ignoresdpversion' option set.
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* The 'nat' option has now been been changed to have yes, no, force_rport, and
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comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
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symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
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remote side requests it and disables symmetric RTP support. Setting it to
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force_rport forces RFC 3581 behavior and disables symmetric RTP support.
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Setting it to comedia enables RFC 3581 behavior if the remote side requests it
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and enables symmetric RTP support.
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* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
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response. This permits the master channel to know how each channel dialled
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in a multi-channel setup resolved in an individual way.
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* Added 'externtcpport' and 'externtlsport' options to allow custom port
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configuration for the externip and externhost options when tcp or tls is used.
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* Added support for message body (stored in content variable) to SIP NOTIFY message
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accessible via AMI and CLI.
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* Added 'media_address' configuration option which can be used to explicitly specify
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the IP address to use in the SDP for media (audio, video, and text) streams.
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* Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
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that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
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received.
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* Added 'use_q850_reason' configuration option for generating and parsing
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if available Reason: Q.850;cause=<cause code> header. It is implemented
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in some gateways for better passing PRI/SS7 cause codes via SIP.
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* When dialing SIP peers, a new component may be added to the end of the dialstring
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to indicate that a specific remote IP address or host should be used when dialing
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the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
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* SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
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ability to selectively force bridged channels to also be encrypted is also
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implemented. Branching in the dialplan can be done based on whether or not
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a channel has secure media and/or signaling.
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* Added directmediapermit/directmediadeny to limit which peers can send direct media
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to each other
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* Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
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Charge messages to snom phones.
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* Added support for G.719 media streams.
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* Added support for 16khz signed linear media streams.
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* SIP is now able to bind to and communicate with IPv6 addresses. In addition,
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RTP has been outfitted with the same abilities.
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* Added support for setting the Max-Forwards: header in SIP requests. Setting is
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available in device configurations as well as in the dial plan.
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* Addition of the 'subscribe_network_change' option for turning on and off
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res_stun_monitor module support in chan_sip.
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* Addition of the 'auth_options_requests' option for turning on and off
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authentication for OPTIONS requests in chan_sip.
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IAX2 Changes
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-----------
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* Added rtsavesysname option into iax.conf to allow the systname to be saved
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on realtime updates.
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* Added the ability for chan_iax2 to inform the dialplan whether or not
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encryption is being used. This interoperates with the SIP SRTP implementation
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so that a secure SIP call can be bridged to a secure IAX call when the
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dialplan requires bridged channels to be "secure".
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* Addition of the 'subscribe_network_change' option for turning on and off
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res_stun_monitor module support in chan_iax.
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MGCP Changes
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------------
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* Added ability to preset channel variables on indicated lines with the setvar
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configuration option. Also, clearvars=all resets the list of variables back
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to none.
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* PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
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See configs/res_pktccops.conf for more information.
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XMPP Google Talk/Jingle changes
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-------------------------------
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* Added the externip option to gtalk.conf.
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* Added the stunaddr option to gtalk.conf which allows for the automatic
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retrieval of the external ip from a stun server.
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Applications
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------------
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* Added 'p' option to PickupChan() to allow for picking up channel by the first
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match to a partial channel name.
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* Added .m3u support for Mp3Player application.
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* Added progress option to the app_dial D() option. When progress DTMF is
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present, those values are sent immediately upon receiving a PROGRESS message
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regardless if the call has been answered or not.
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* Added functionality to the app_dial F() option to continue with execution
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at the current location when no parameters are provided.
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* Added the 'a' option to app_dial to answer the calling channel before any
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announcements or macros are executed.
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* Modified app_dial to set answertime when the called channel answers even if
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the called channel hangs up during playback of an announcement.
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* Modified app_dial 'r' option to support an additional parameter to play an
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indication tone from indications.conf
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* Added c() option to app_chanspy. This option allows custom DTMF to be set
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to cycle through the next available channel. By default this is still '*'.
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* Added x() option to app_chanspy. This option allows DTMF to be set to
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exit the application.
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* The Voicemail application has been improved to automatically ignore messages
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that only contain silence.
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* If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
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associated mailbox(es) to be greetings-only.
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* The ChanSpy application now has the 'S' option, which makes the application
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automatically exit once it hits a point where no more channels are available
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to spy on.
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* The ChanSpy application also now has the 'E' option, which spies on a single
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channel and exits when that channel hangs up.
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* The MeetMe application now turns on the DENOISE() function by default, for
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each participant. In our tests, this has significantly decreased background
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noise (especially noisy data centers).
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* Voicemail now permits storage of secrets in a separate file, located in the
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spool directory of each individual user. The control for this is located in
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the "passwordlocation" option in voicemail.conf. Please see the sample
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configuration for more information.
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* The ChanIsAvail application now exposes the returned cause code using a separate
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variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
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* Added 'd' option to app_followme. This option disables the "Please hold"
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announcement.
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* Added 'y' option to app_record. This option enables a mode where any DTMF digit
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received will terminate recording.
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* Voicemail now supports per mailbox settings for folders when using IMAP storage.
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Previously the folder could only be set per context, but has now been extended
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using the imapfolder option.
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* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
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* Voicemail now allows the pager date format to be specified separately from the
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email date format.
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* New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
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to allow joining, leaving, and sending text to group chats.
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* MeetMe has a new option 'G' to play an announcement before joining a conference.
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* Page has a new option 'A(x)' which will playback an announcement simultaneously
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to all paged phones (and optionally excluding the caller's one using the new
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option 'n') before the call is bridged.
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* The 'f' option to Dial has been augmented to take an optional argument. If no
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argument is provided, the 'f' option works as it always has. If an argument is
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provided, then the connected party information of all outgoing channels created
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during the Dial will be set to the argument passed to the 'f' option.
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* Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
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Gosub on the peer.
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* The OSP lookup application adds in/outbound network ID, optional security,
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number portability, QoS reporting, destination IP port, custom info and service
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type features.
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* Added new application VMSayName that will play the recorded name of the voicemail
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user if it exists, otherwise will play the mailbox number.
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* Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
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retrieve state for a particular bridge, where <name> is the conference name
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* app_directory now allows exiting at any time using the operator or pound key.
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* Voicemail now supports setting a locale per-mailbox.
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* Two new applications are provided for declining counting phrases in multiple
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languages. See the application notes for SayCountedNoun and SayCountedAdj for
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more information.
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* Voicemail now runs the externnotify script when pollmailboxes is activated and
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notices a change.
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* Voicemail now includes rdnis within msgXXXX.txt file.
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* Added 'D' command to ExternalIVR full details in doc/externalivr.txt
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* Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
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a MeetMe conference
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Dialplan Functions
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------------------
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* SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
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over SRV records associated with a specific service. From the CLI, type
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'core show function SRVQUERY' and 'core show function SRVRESULT' for more
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details on how these may be used.
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* PITCH_SHIFT dialplan function added. This function can be used to modify the
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pitch of a channel's tx and rx audio streams.
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* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
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setting various connected line and redirecting party information.
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* CALLERID and CONNECTEDLINE dialplan functions have been extended to
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support ISDN subaddressing.
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* The CHANNEL() function now supports the "name" and "checkhangup" options.
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* For DAHDI channels, the CHANNEL() dialplan function now allows
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the dialplan to request changes in the configuration of the active
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echo canceller on the channel (if any), for the current call only.
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The syntax is:
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exten => s,n,Set(CHANNEL(echocan_mode)=off)
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The possible values are:
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on - normal mode (the echo canceller is actually reinitialized)
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off - disabled
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fax - FAX/data mode (NLP disabled if possible, otherwise completely
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disabled)
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voice - voice mode (returns from FAX mode, reverting the changes that
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were made when FAX mode was requested)
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* Added new dialplan function MASTER_CHANNEL(), which permits retrieving
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and setting variables on the channel which created the current channel.
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Administrators should take care to avoid naming conflicts, when multiple
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channels are dialled at once, especially when used with the Local channel
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construct (which all could set variables on the master channel). Usage
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of the HASH() dialplan function, with the key set to the name of the slave
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channel, is one approach that will avoid conflicts.
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* Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
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audio in a channel.
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* func_odbc now allows multiple row results to be retrieved without using
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mode=multirow. If rowlimit is set, then additional rows may be retrieved
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from the same query by using the name of the function which retrieved the
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first row as an argument to ODBC_FETCH().
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* Added JABBER_RECEIVE, which permits receiving XMPP messages from the
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dialplan. This function returns the content of the received message.
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* Added REPLACE, which searches a given variable name for a set of characters,
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then either replaces them with a single character or deletes them.
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* Added PASSTHRU, which literally passes the same argument back as its return
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value. The intent is to be able to use a literal string argument to
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functions that currently require a variable name as an argument.
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* HASH-associated variables now can be inherited across channel creation, by
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prefixing the name of the hash at assignment with the appropriate number of
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underscores, just like variables.
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* GROUP_MATCH_COUNT has been improved to allow regex matching on category
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* CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
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whether or not channels that are bridged to the current channel will be
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required to have secure signaling and/or media.
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* CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
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the current channel has secure signaling and/or media.
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* For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
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"no_media_path" option.
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Returns "0" if there is a B channel associated with the call.
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Returns "1" if no B channel is associated with the call. The call is either
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on hold or is a call waiting call.
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* Added option to dialplan function CDR(), the 'f' option
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allows for high resolution times for billsec and duration fields.
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* FILE() now supports line-mode and writing.
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* Added FIELDNUM(), which returns the 1-based offset of a field in a list.
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* FRAME_TRACE(), for tracking internal ast_frames on a channel.
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Dialplan Variables
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------------------
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* Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
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* Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
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and is set when a dynamic feature is triggered.
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* Added PARKINGLOT which can be used with parkeddynamic feature.conf option
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to dynamically create a new parking lot matching the value this varible is
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set to.
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* Added PARKINGDYNAMIC which represents the template parkinglot defined in
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features.conf that should be the base for dynamic parkinglots.
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* Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
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parkinglot should have.
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* Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
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should have.
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Queue changes
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-------------
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* Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
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timeout has expired.
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* Added 'R' option to app_queue. This option stops moh and indicates ringing
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to the caller when an Agent's phone is ringing. This can be used to indicate
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to the caller that their call is about to be picked up, which is nice when
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one has been on hold for an extened period of time.
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* A new config option, penaltymemberslimit, has been added to queues.conf.
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When set this option will disregard penalty settings when a queue has too
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few members.
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* A new option, 'I' has been added to both app_queue and app_dial.
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By setting this option, Asterisk will not update the caller with
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connected line changes or redirecting party changes when they occur.
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* A 'relative-peroidic-announce' option has been added to queues.conf. When
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enabled, this option will cause periodic announce times to be calculated
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from the end of announcements rather than from the beginning.
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* The autopause option in queues.conf can be passed a new value, "all." The
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result is that if a member becomes auto-paused, he will be paused in all
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queues for which he is a member, not just the queue that failed to reach
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the member.
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* Added dialplan function QUEUE_EXISTS to check if a queue exists
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* The queue logger now allows events to optionally propagate to a file,
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even when realtime logging is turned on. Additionally, realtime logging
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supports sending the event arguments to 5 individual fields, although it
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will fallback to the previous data definition, if the new table layout is
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not found.
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mISDN channel driver (chan_misdn) changes
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----------------------------------------
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* Added display_connected parameter to misdn.conf to put a display string
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in the CONNECT message containing the connected name and/or number if
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the presentation setting permits it.
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* Added display_setup parameter to misdn.conf to put a display string
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in the SETUP message containing the caller name and/or number if the
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presentation setting permits it.
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* Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
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indicate the dialplan settings are to be obtained from the asterisk
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channel.
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* Made misdn.conf parameter callerid accept the "name" <number> format
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used by the rest of the system.
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* Made use the nationalprefix and internationalprefix misdn.conf
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parameters to prefix any received number from the ISDN link if that
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number has the corresponding Type-Of-Number. NOTE: This includes
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comparing the incoming call's dialed number against the MSN list.
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* Added the following new parameters: unknownprefix, netspecificprefix,
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subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
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received number from the ISDN link if that number has the corresponding
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Type-Of-Number.
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* Added new dialplan application misdn_command which permits controlling
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the CCBS/CCNR functionality.
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* Added new dialplan function mISDN_CC which permits retrieval of various
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values from an active call completion record.
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* For PTP, you should manually send the COLR of the redirected-to party
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for an incomming redirected call if the incoming call could experience
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further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
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set the REDIRECTING(to-pres) to the COLR. A call has been redirected
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if the REDIRECTING(from-num) is not empty.
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* For outgoing PTP redirected calls, you now need to use the inhibit(i)
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option on all of the REDIRECTING statements before dialing the
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redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
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and the REDIRECTING(from-xxx,i) values. The PTP call will update the
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redirecting-to presentation (COLR) when it becomes available.
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* Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
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information.
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thirdparty mISDN enhancements
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-----------------------------
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mISDN has been modified by Digium, Inc. to greatly expand facility message
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support to allow:
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* Enhanced COLP support for call diversion and transfer.
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* CCBS/CCNR support.
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The latest modified mISDN v1.1.x based version is available at:
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http://svn.digium.com/svn/thirdparty/mISDN/trunk
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http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
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Tagged versions of the modified mISDN code are available under:
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http://svn.digium.com/svn/thirdparty/mISDN/tags
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http://svn.digium.com/svn/thirdparty/mISDNuser/tags
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libpri channel driver (chan_dahdi) DAHDI changes
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|
-------------------------------------------
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* The channel variable PRIREDIRECTREASON is now just a status variable
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and it is also deprecated. Use the REDIRECTING(reason) dialplan function
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to read and alter the reason.
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* For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
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redirected-to party for an incomming redirected call if the incoming call
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could experience further redirects. Just set the
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REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
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to the COLR. A call has been redirected if the REDIRECTING(count) is not
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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.
|
|
|
|
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 doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) 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
|
|
"Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
|
|
|
|
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 doc/distributed_devstate-XMPP.txt
|
|
* 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.
|
|
|
|
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, see doc/distributed_devstate.txt.
|
|
|
|
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 doc/externalivr.txt for the
|
|
complete documentation.
|
|
* 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 doc/hoard.txt
|
|
for more information.
|
|
* 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
|
|
in doc/manager_1_1.txt
|
|
* 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 doc/manager_1_1.txt for more details or
|
|
manager show command Atxfer from the CLI
|
|
* New command: IAXregistry. See doc/manager_1_1.txt 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 doc/siptls.txt 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 doc/unistim.txt 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 in doc/qos.tex, or the IP Quality of Service
|
|
section of asterisk.pdf.
|
|
* 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.
|