====================================================================== === === This file documents the new and/or enhanced functionality added in === the Asterisk versions listed below. This file does NOT include === changes in behavior that would not be backwards compatible with === previous versions; for that information see the UPGRADE.txt file === and the other UPGRADE files for older releases. === ====================================================================== SIP changes ----------- * Added a new option "prematuremedia" that defaults to "no". If you turn this option on, chan_sip will not automatically initiate early media if it receives audio from the incoming channel before there's been a progress indication. ---------------------------------------------------------------------------------- --- Functionality changes from Asterisk 1.6.1.1 to Asterisk 1.6.1.2 ------------- ---------------------------------------------------------------------------------- SIP Changes ----------- * 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. In addition, this behavior is automatic when the SDP received is in response to a T.38 re-INVITE that Asterisk initiated. In this situation, since the call will fail if Asterisk does not process the incoming SDP, Asterisk will accept the SDP even if the SDP version number is not properly incremented, but will generate a warning in the log indicating that the SIP peer that sent the SDP should have the 'ignoresdpversion' option set. ------------------------------------------------------------------------------ --- 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. The dial() application ---------------------- * 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. * 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)}/" . The chanspy() application ------------------------- * 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. * 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. Other 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 * The voicemail externnotify script now accepts an additional (last) parameter containing the number of urgent messages in the INBOX. * The Jack application now has a c() option to supply a custom client name. * 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. * 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. Initially, we saw 4x improvement in call setup/destruction, but at the time of merging, this gain has disappeared; further research will be done to try and restore this performance improvement. Astobj2 refcounting is now used for users, peers, and dialogs. Users are encouraged to assist in regression testing and problem reporting! * Added ability to specify registration expiry time on a per registration basis in the register line. * Added support for Realtime Text redundancy - T140 RED - 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, "config reload " 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 action now takes an optional list of variables to display along with channel status. 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 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 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. ------------------------------------------------------------------------------ --- 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 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 ----------- * New CLI command "core show hint" (usage: core show hint ) * 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. 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. * Improved NAT and STUN support. chan_sip now can use port numbers in bindaddr, externip and externhost options, as well as contact a STUN server to detect its external address for the SIP socket. See sip.conf.sample, 'NAT' section. * 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. 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 * 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 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. 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.