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https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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======================================================================
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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 1.6.0 to Asterisk 1.6.1 -------------
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------------------------------------------------------------------------------
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Device State Handling
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---------------------
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* The event infrastructure in Asterisk got another big update to help support
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distributed events. It currently supports distributed device state and
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distributed Voicemail MWI (Message Waiting Indication). A new module has
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been merged, res_ais, which facilitates communicating events between servers.
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It uses the SAForum AIS (Service Availability Forum Application Interface
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Specification) CLM (Cluster Management) and EVT (Event) services to maintain
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a cluster of Asterisk servers, and to share events between them. For more
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information on setting this up, see doc/distributed_devstate.txt.
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Dialplan Functions
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------------------
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* Added a new dialplan function, AST_CONFIG(), which allows you to access
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variables from an Asterisk configuration file.
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* The JACK_HOOK function now has a c() option to supply a custom client name.
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* Added two new dialplan functions from libspeex for audio gain control and
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denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
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rx directions of a channel from the dialplan.
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* The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
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based on other parameters. The default is still to search based on the
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forwarding station ID. However, there are new options that allow you to search
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based on the message desk terminal ID, or the message desk number.
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* TIMEOUT() has been modified to be accurate down to the millisecond.
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* ENUM*() functions now include the following new options:
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- 'u' returns the full URI and does not strip off the URI-scheme.
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- 's' triggers ISN specific rewriting
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- 'i' looks for branches into an Infrastructure ENUM tree
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- 'd' for a direct DNS lookup without any flipping of digits.
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* TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
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* CHANNEL() now has options for the maximum, minimum, and standard or normal
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deviation of jitter, rtt, and loss for a call using chan_sip.
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DAHDI channel driver (chan_dahdi) Changes
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----------------------------------------
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* Channels can now be configured using named sections in chan_dahdi.conf, just
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like other channel drivers, including the use of templates.
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* The default for pridialplan has changed from 'national' to 'unknown'.
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PBX Changes
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-----------
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* It is now possible to specify a pattern match as a hint. Once a phone subscribes
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to something that matches the pattern a hint will be created using the contents
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and variables evaluated.
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* Dialplan matching has been extended to allow an extension to return to the
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PBX core to wait for more digits. This is done by using the new dialplan
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application called "Incomplete". This will permit a whole new level of
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extension control, by giving the administrator more control over early
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matches employing one of the short-circuit pattern match operators. Note
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that custom applications can trigger this same behavior by returning the
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special value AST_PBX_INCOMPLETE.
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Application Changes
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-------------------
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* Directory now permits both first and last names to be matched at the same
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time. In addition, the number of digits to enter of the name can be set in
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the arguments to Directory; previously, you could enter only 3, regardless
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of how many names are in your company. For large companies, this should be
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quite helpful.
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* Voicemail now permits a mailbox setting to wrap around from first to last
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messages, if the "messagewrap" option is set to a true value.
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* Voicemail now permits an external script to be run, for password validation.
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The script should output "VALID" or "INVALID" on stdout, depending upon the
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wish to validate or invalidate the password given. Arguments are:
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"mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
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more details
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* The voicemail externnotify script now accepts an additional (last) parameter
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containing the number of urgent messages in the INBOX.
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* Dial has a new option: F(context^extension^pri), which permits a callee to
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continue in the dialplan, at the specified label, if the caller hangs up.
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* ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
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technology name (e.g. SIP, IAX, etc) of the channel being spied on.
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* The Jack application now has a c() option to supply a custom client name.
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* Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
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like the pre-existing whisper mode, except that the spy can also talk to the
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participant on the bridged channel as well.
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* Chanspy has a new option, 'n', which will allow for the spied-on party's name
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to be spoken instead of the channel name or number. For more information on the
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use of this option, issue the command "core show application ChanSpy" from the
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Asterisk CLI.
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* Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
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spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
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words, if using the 'd' option, it is not possible to enter a number to append to
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the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
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change to whisper mode, and pressing 6 will change to barge mode.
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* ExternalIVR now takes several options that affect the way it performs, as
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well as having several new commands. Please see doc/externalivr.txt for the
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complete documentation.
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* Added ability to communicate over a TCP socket instead of forking a child process for the
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ExternalIVR application.
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* ChanIsAvail has a new option, 'a', which will return all available channels instead
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of just the first one if you give the function more then one channel to check.
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* PrivacyManager now takes an option where you can specify a context where the
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given number will be matched. This way you have more control over who is allowed
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and it stops the people who blindly enter 10 digits.
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* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
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answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
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from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
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original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
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the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
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obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
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* The Dial() application no longer copies the language used by the caller to the callee's
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channel. If you desire for the caller's channel's language to be used for file playback
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to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
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* SendImage() no longer hangs up the channel on error; instead, it sets the
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status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
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'UNSUPPORTED'. This change makes SendImage() more consistent with other
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applications.
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* Park has a new option, 's', which silences the announcement of the parking space number.
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* A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
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invalid input and will be assumed to mean that no timeout is desired.
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SIP Changes
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-----------
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* Added DNS manager support to registrations for peers referencing peer entries.
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DNS manager runs in the background which allows DNS lookups to be run asynchronously
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as well as periodically updating the IP address. These properties allow for
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better performance as well as recovery in the event of an IP change.
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* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
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load/reload of large numbers of peers/users by ~40x (for large lists of peers.
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Initially, we saw 4x improvement in call setup/destruction, but at the time
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of merging, this gain has disappeared; further research will be done to try
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and restore this performance improvement. Astobj2 refcounting is now used
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for users, peers, and dialogs. Users are encouraged to assist in regression
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testing and problem reporting!
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* Added ability to specify registration expiry time on a per registration basis in
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the register line.
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* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
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lost packets.
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* Added t38pt_usertpsource option. See sip.conf.sample for details.
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* Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
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* 'sip show peers' and 'sip show users' display their entries sorted in
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alphabetical order, as opposed to the order they were in, in the config
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file or database.
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* Videosupport now supports an additional option, "always", which always sets
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up video RTP ports, even on clients that don't support it. This helps with
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callfiles and certain transfers to ensure that if two video phones are
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connected, they will always share video feeds.
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IAX Changes
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-----------
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* Existing DNS manager lookups extended to check for SRV records.
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* IAX2 encryption support has been improved to support periodic key rotation
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within a call for enhanced security. The option "keyrotate" has been
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provided to disable this functionality to preserve backwards compatibility
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with older versions of IAX2 that do not support key rotation.
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CLI Changes
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-----------
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* New CLI command, "config reload <file.conf>" which reloads any module that
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references that particular configuration file. Also added "config list"
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which shows which configuration files are in use.
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* New CLI commands, "pri show version" and "ss7 show version" that will
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display which version of libpri and libss7 are being used, respectively.
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A new API call was added so trunk will now have to be compiled against
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a versions of libpri and libss7 that have them or it will not know that
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these libraries exist.
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* The commands "core show globals", "core set global" and "core set chanvar" has
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been deprecated in favor of the more semanticly correct "dialplan show globals",
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"dialplan set chanvar" and "dialplan set global".
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* New CLI command "dialplan show chanvar" to list all variables associated
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with a given channel.
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DNS manager changes
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-------------------
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* Addresses managed by DNS manager now can check to see if there is a DNS
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SRV record for a given domain and will use that hostname/port if present.
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AMI - The manager (TCP/TLS/HTTP)
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--------------------------------
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* The Status command now takes an optional list of variables to display
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along with channel status.
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ODBC Changes
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------------
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* res_odbc no longer has a limit of 1023 total possible unshared connections,
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as some people were running into this limit. This limit has been increased
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to 4.2 billion.
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Queue changes
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-------------
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* The TRANSFER queue log entry now includes the the caller's original
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position in the transferred-from queue.
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* A new configuration option, "timeoutpriority" has been added. Please see the section labeled
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"QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
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as well as an explanation about timeout options in general
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Realtime changes
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----------------
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* Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
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adaptive capabilities. What this means in practical terms is that if your
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realtime table lacks critical fields, Asterisk will now emit warnings to
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that effect. Also, some of the realtime drivers have the ability (if
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configured) to automatically add those columns to the table with the
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correct type and length.
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Miscellaneous
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-------------
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* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
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the 'setvar' option to cause a given audio file to be played upon completion
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of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
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Skinny channels only.
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* You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
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for more information.
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* Config file variables may now be appended to, by using the '+=' append
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operator. This is most helpful when working with long SQL queries in
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func_odbc.conf, as the queries no longer need to be specified on a single
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line.
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
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------------------------------------------------------------------------------
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AMI - The manager (TCP/TLS/HTTP)
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--------------------------------
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* Manager has undergone a lot of changes, all of them documented
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in doc/manager_1_1.txt
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* Manager version has changed to 1.1
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* Added a new action 'CoreShowChannels' to list currently defined channels
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and some information about them.
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* Added a new action 'SIPshowregistry' to list SIP registrations.
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* Added TLS support for the manager interface and HTTP server
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* Added the URI redirect option for the built-in HTTP server
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* The output of CallerID in Manager events is now more consistent.
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CallerIDNum is used for number and CallerIDName for name.
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* Enable https support for builtin web server.
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See configs/http.conf.sample for details.
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* Added a new action, GetConfigJSON, which can return the contents of an
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Asterisk configuration file in JSON format. This is intended to help
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improve the performance of AJAX applications using the manager interface
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over HTTP.
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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indicate channel driver. Previously, we used a mixture of "Channel"
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and "ChannelDriver" headers.
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* Added a "Bridge" action which allows you to bridge any two channels that
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are currently active on the system.
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* Added a "ListAllVoicemailUsers" action that allows you to get a list of all
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the voicemail users setup.
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* Added 'DBDel' and 'DBDelTree' manager commands.
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* cdr_manager now reports events via the "cdr" level, separating it from
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the very verbose "call" level.
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* Manager users are now stored in memory. If you change the manager account
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list (delete or add accounts) you need to reload manager.
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* Added Masquerade manager event for when a masquerade happens between
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two channels.
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* Added "manager reload" command for the CLI
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* Lots of commands that only provided information are now allowed under the
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Reporting privilege, instead of only under Call or System.
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* The IAX* commands now require either System or Reporting privilege, to
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mirror the privileges of the SIP* commands.
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* Added ability to retrieve list of categories in a config file.
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* Added ability to retrieve the content of a particular category.
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* Added ability to empty a context.
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* Created new action to create a new file.
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* Updated delete action to allow deletion by line number with respect to category.
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* Added new action insert to add new variable to category at specified line.
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* Updated action newcat to allow new category to be inserted in file above another
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existing category.
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* Added new event "JitterBufStats" in the IAX2 channel
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* Originate now requires the Originate privilege and, if you want to call out
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to a subshell, it requires the System privilege, as well. This was done to
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enhance manager security.
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* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
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* New command: Atxfer. See doc/manager_1_1.txt for more details or
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manager show command Atxfer from the CLI
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Dialplan functions
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------------------
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* Added the DEVICE_STATE() dialplan function which allows retrieving any device
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state in the dialplan, as well as creating custom device states that are
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controllable from the dialplan.
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* Extend CALLERID() function with "pres" and "ton" parameters to
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fetch string representation of calling number presentation indicator
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and numeric representation of type of calling number value.
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* MailboxExists converted to dialplan function
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* A new option to Dial() for telling IP phones not to count the call
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as "missed" when dial times out and cancels.
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* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
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mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
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held for any given channel. Also, locks are automatically freed when a
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channel is hung up.
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* Added HINT() dialplan function that allows retrieving hint information.
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Hints are mappings between extensions and devices for the sake of
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determining the state of an extension. This function can retrieve the list
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of devices or the name associated with a hint.
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* Added EXTENSION_STATE() dialplan function which allows retrieving the state
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of any extension.
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* Added SYSINFO() dialplan function which allows retrieval of system information
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* Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
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the existence of a dialplan target.
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* Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
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upper and lower case, respectively.
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* When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
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ID for the call (not the Asterisk call ID or unique ID), provided that the
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channel driver supports this. For SIP, you get the SIP call-ID for the
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bridged channel which you can store in the CDR with a custom field.
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CLI Changes
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-----------
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* New CLI command "core show hint" (usage: core show hint <exten>)
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* New CLI command "core show settings"
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* Added 'core show channels count' CLI command.
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* Added the ability to set the core debug and verbose values on a per-file basis.
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* Added 'queue pause member' and 'queue unpause member' CLI commands
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* Ability to set process limits ("ulimit") without restarting Asterisk
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* Enhanced "agi debug" to print the channel name as a prefix to the debug
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output to make debugging on busy systems much easier.
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* New CLI commands "dialplan set extenpatternmatching true/false"
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* New CLI command: "core set chanvar" to set a channel variable from the CLI.
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* Added an easy way to execute Asterisk CLI commands at startup. Any commands
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listed in the startup_commands section of cli.conf will get executed.
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* Added a CLI command, "devstate change", which allows you to set custom device
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states from the func_devstate module that provides the DEVICE_STATE() function
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and handling of the "Custom:" devices.
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* New CLI command: "sip show sched" which shows all ast_sched entries for sip,
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sorted into the different possible callbacks, with the number of entries
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currently scheduled for each. Gives you a feel for how busy the sip channel
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driver is.
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SIP changes
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-----------
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* Improved NAT and STUN support.
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chan_sip now can use port numbers in bindaddr, externip and externhost
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options, as well as contact a STUN server to detect its external address
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for the SIP socket. See sip.conf.sample, 'NAT' section.
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* The default SIP useragent= identifier now includes the Asterisk version
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from the Digest header
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rather than from the From: field. This feature is considered experimental.
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* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
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since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
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* The "localmask" setting was removed in version 1.2 and the reminder about it
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being removed is now also removed.
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* A new option "busylevel" for setting a level of calls where asterisk reports
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a device as busy, to separate it from call-limit. This value is also added
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to the SIP_PEER dialplan function.
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* A new realtime family called "sipregs" is now supported to store SIP registration
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data. If this family is defined, "sippeers" will be used for configuration and
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"sipregs" for registrations. If it's not defined, "sippeers" will be used for
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registration data, as before.
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* The SIPPEER function have new options for port address, call and pickup groups
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* Added support for T.140 realtime text in SIP/RTP
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* The "checkmwi" option has been removed from sip.conf, as it is no longer
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required due to the restructuring of how MWI is handled. See the descriptions
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in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
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for more information.
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* Added rtpdest option to CHANNEL() dialplan function.
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* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
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* SIP now adds a header to the CANCEL if the call was answered by another phone
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in the same dial command, or if the new c option in dial() is used.
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* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
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states it is not needed. For phones, however, that do require it the "registertrying" option
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has been added so it can be enabled.
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* A new option called "callcounter" (global/peer/user level) enables call counters needed
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for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
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used to enable this functionality).
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* New settings for timer T1 and timer B on a global level or per device. This makes it
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possible to force timeout faster on non-responsive SIP servers. These settings are
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considered advanced, so don't use them unless you have a problem.
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* Added a dial string option to be able to set the To: header in an INVITE to any
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SIP uri.
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* Added a new global and per-peer option, qualifyfreq, which allows you to configure
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the qualify frequency.
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* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
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were not properly torn down due to network or endpoint failures during an established
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SIP session.
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* Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
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configs/sip.conf.sample for more information on how it is used.
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* Added a new configuration option "authfailureevents" that enables manager events when
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a peer can't authenticate properly.
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* Added DNS manager support to registrations for peers not referencing a peer entry.
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|
|
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.
|
|
|
|
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.
|
|
* 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.
|
|
|
|
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.
|
|
|
|
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 '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
|
|
* 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.
|
|
|
|
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.
|
|
|
|
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 logfiles,
|
|
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
|
|
rotatestrategry=rotate, to allow a limit on the number of logfiles 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.
|