UPGRADE.txt

=========================================================
=== Information for upgrading from Asterisk 1.2 to 1.4
===
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE.txt     -- Upgrade info for 1.2 to 1.4
=========================================================

IAX2:

* The firmware for the IAXy has been removed from Asterisk.  It can be
  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
  install the firmware into its proper location, place the firmware in the
  contrib/firmware/iax/ directory in the Asterisk source tree before running
  "make install".

* There have been some changes to the IAX2 protocol to address the security
  concerns documented in the security advisory AST-2009-006.  Please see the
  IAX2 security document, doc/IAX2-security.pdf, for information regarding
  backwards compatibility with versions of Asterisk that do not contain these
  changes to IAX2.

Build Process (configure script):

Asterisk now uses an autoconf-generated configuration script to learn how it
should build itself for your system. As it is a standard script, running:

$ ./configure --help

will show you all the options available. This script can be used to tell the
build process what libraries you have on your system (if it cannot find them
automatically), which libraries you wish to have ignored even though they may
be present, etc.

You must run the configure script before Asterisk will build, although it will
attempt to automatically run it for you with no options specified; for most
users, that will result in a similar build to what they would have had before
the configure script was added to the build process (except for having to run
'make' again after the configure script is run). Note that the configure script
does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
when your system configuration changes or you wish to build Asterisk with 
different options.

Build Process (module selection):

The Asterisk source tree now includes a basic module selection and build option
selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
In this tool, you can disable building of modules that you don't care about,
turn on/off global options for the build and see which modules will not 
(and cannot) be built because your system does not have the required external
dependencies installed.

The resulting file from menuselect is called 'menuselect.makeopts'. Note that
the resulting menuselect.makeopts file generally contains which modules *not*
to build. The modules listed in this file indicate which modules have unmet
dependencies, a present conflict, or have been disabled by the user in the
menuselect interface. Compiler Flags can also be set in the menuselect
interface.  In this case, the resulting file contains which CFLAGS are in use,
not which ones are not in use.

If you would like to save your choices and have them applied against all
builds, the file can be copied to '~/.asterisk.makeopts' or 
'/etc/asterisk.makeopts'.

Build Process (Makefile targets):

The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
in the menuselect tool.

It is now possible to run most make targets against a single subdirectory; from
the top level directory, for example, 'make channels' will run 'make all' in the
'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.

Sound (prompt) and Music On Hold files:

Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
use with Asterisk have been replaced with new versions produced from high quality
master recordings, and are available in three languages (English, French and
Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
In addition, the music on hold files provided by opsound.org Music are now available
in the same five formats, but no longer available in MP3 format.

The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
All of the other variations can be installed by running 'make menuselect' and
selecting the packages you wish to install; when you run 'make install', those
packages will be downloaded and installed along with the standard files included
in the tarball.

If for some reason you expect to not have Internet access at the time you will be
running 'make install', you can make your package selections using menuselect and
then run 'make sounds' to download (only) the sound packages; this will leave the
sound packages in the 'sounds' subdirectory to be used later during installation.

WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
instead of the alternate-language files being stored in subdirectories underneath
the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
language itself, then places all the sound files for that language under that
directory and its subdirectories. This is the layout that will be created if you
select non-English languages to be installed via menuselect, HOWEVER Asterisk does
not default to this layout and will not find the files in the places it expects them
to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
installed.

PBX Core:

* The (very old and undocumented) ability to use BYEXTENSION for dialing
  instead of ${EXTEN} has been removed.
  
* Builtin (res_features) transfer functionality attempts to use the context
  defined in TRANSFER_CONTEXT variable of the transferer channel first. If
  not set, it uses the transferee variable. If not set in any channel, it will 
  attempt to use the last non macro context. If not possible, it will default
  to the current context.

* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
  if your dialplan relies on the ability to 'run off the end' of an extension
  and wait for a new extension without using WaitExten() to accomplish that,
  you will need set autofallthrough to 'no' in your extensions.conf file.

Language Support:

* Support for Taiwanese was incorrectly supported with the "tw" language code.
  In reality, the "tw" language code is reserved for the Twi language, native
  to Ghana.  If you were previously using the "tw" language code, you should
  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
  specific localizations.  Additionally, "mx" should be changed to "es_MX",
  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
  "cs", not "cz".
 
Command Line Interface:

* 'show channels concise', designed to be used by applications that will parse
  its output, previously used ':' characters to separate fields. However, some
  of those fields can easily contain that character, making the output not
  parseable. The delimiter has been changed to '!'.

Applications:

* In previous Asterisk releases, many applications would jump to priority n+101
  to indicate some kind of status or error condition.  This functionality was
  marked deprecated in Asterisk 1.2.  An option to disable it was provided with
  the default value set to 'on'.  The default value for the global priority
  jumping option is now 'off'.

* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
  AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
  and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
  been removed in this version.  You should use the equivalent dialplan
  function in places where you have previously used one of these applications.

* The application SetGlobalVar has been deprecated.  You should replace uses
  of this application with the following combination of Set and GLOBAL():
  Set(GLOBAL(name)=value).  You may also access global variables exclusively by
  using the GLOBAL() dialplan function, instead of relying on variable
  interpolation falling back to globals when no channel variable is set.

* The application SetVar has been renamed to Set.  The syntax SetVar was marked
  deprecated in version 1.2 and is no longer recognized in this version.  The
  use of Set with multiple argument pairs has also been deprecated.  Please
  separate each name/value pair into its own dialplan line.

* app_read has been updated to use the newer options codes, using "skip" or
  "noanswer" will not work.  Use s or n.  Also there is a new feature i, for
  using indication tones, so typing in skip would give you unexpected results.

* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.

* The CONNECT event in the queue_log from app_queue now has a second field 
  in addition to the holdtime field. It contains the unique ID of the 
  queue member channel that is taking the call. This is useful when trying 
  to link recording filenames back to a particular call from the queue.  

* The old/current behavior of app_queue has a serial type behavior
  in that the queue will make all waiting callers wait in the queue
  even if there is more than one available member ready to take
  calls until the head caller is connected with the member they
  were trying to get to. The next waiting caller in line then
  becomes the head caller, and they are then connected with the
  next available member and all available members and waiting callers
  waits while this happens. This cycle continues until there are
  no more available members or waiting callers, whichever comes first.
  The new behavior, enabled by setting autofill=yes in queues.conf
  either at the [general] level to default for all queues or 
  to set on a per-queue level, makes sure that when the waiting 
  callers are connecting with available members in a parallel fashion 
  until there are no more available members or no more waiting callers,
  whichever comes first. This is probably more along the lines of how
  one would expect a queue should work and in most cases, you will want 
  to enable this new behavior. If you do not specify or comment out this 
  option, it will default to "no" to keep backward compatability with the old 
  behavior.

* Queues depend on the channel driver reporting the proper state
  for each member of the queue. To get proper signalling on
  queue members that use the SIP channel driver, you need to
  enable a call limit (could be set to a high value so it
  is not put into action) and also make sure that both inbound
  and outbound calls are accounted for.

  Example:

       [general]
       limitonpeer = yes

       [peername]
       type=friend
       call-limit=10


* The app_queue application now has the ability to use MixMonitor to 
  record conversations queue members are having with queue callers. Please
  see configs/queues.conf.sample for more information on this option.

* The app_queue application strategy called 'roundrobin' has been deprecated
  for this release. Users are encouraged to use 'rrmemory' instead, since it
  provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
  'rrmemory' will be renamed 'roundrobin'.

* The app_queue application option called 'monitor-join' has been deprecated
  for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
  since it provides the same functionality but is not dependent on soxmix or some
  other external program in order to mix the audio.

* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
  the 'm' option now provides the functionality of "initially muted". 
  In practice, most existing dialplans using the 'm' flag should not notice
  any difference, unless the keypad menu is enabled, allowing the user 
  to unmute themsleves.

* ast_play_and_record would attempt to cancel the recording if a DTMF
  '0' was received.  This behavior was not documented in most of the
  applications that used ast_play_and_record and the return codes from
  ast_play_and_record weren't checked for properly.
  ast_play_and_record has been changed so that '0' no longer cancels a
  recording.  If you want to allow DTMF digits to cancel an
  in-progress recording use ast_play_and_record_full which allows you
  to specify which DTMF digits can be used to accept a recording and
  which digits can be used to cancel a recording.

* ast_app_messagecount has been renamed to ast_app_inboxcount.  There is now a
  new ast_app_messagecount function which takes a single context/mailbox/folder
  mailbox specification and returns the message count for that folder only.
  This addresses the deficiency of not being able to count the number of
  messages in folders other than INBOX and Old.

* The exit behavior of the AGI applications has changed. Previously, when
  a connection to an AGI server failed, the application would cause the channel
  to immediately stop dialplan execution and hangup. Now, the only time that
  the AGI applications will cause the channel to stop dialplan execution is
  when the channel itself requests hangup. The AGI applications now set an
  AGISTATUS variable which will allow you to find out whether running the AGI
  was successful or not.

  Previously, there was no way to handle the case where Asterisk was unable to
  locally execute an AGI script for some reason. In this case, dialplan
  execution will continue as it did before, but the AGISTATUS variable will be
  set to "FAILURE".

  A locally executed AGI script can now exit with a non-zero exit code and this
  failure will be detected by Asterisk. If an AGI script exits with a non-zero
  exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
  "SUCCESS".

* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
  previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
  your table format using the schema provided in doc/odbcstorage.txt

* app_waitforsilence: Fixes have been made to this application which changes the 
  default behavior with how quickly it returns. You can maintain "old-style" behavior
  with the addition/use of a third "timeout" parameter.
  Please consult the application documentation and make changes to your dialplan 
  if appropriate.

Manager:

* After executing the 'status' manager action, the "Status" manager events
  included the header "CallerID:" which was actually only the CallerID number,
  and not the full CallerID string.  This header has been renamed to
  "CallerIDNum".  For compatibility purposes, the CallerID parameter will remain
  until after the release of 1.4, when it will be removed.  Please use the time
  during the 1.4 release to make this transition.

* The AgentConnect event now has an additional field called "BridgedChannel" 
  which contains the unique ID of the queue member channel that is taking the 
  call. This is useful when trying to link recording filenames back to 
  a particular call from the queue.

* app_userevent has been modified to always send Event: UserEvent with the
  additional header UserEvent: <userspec>.  Also, the Channel and UniqueID
  headers are not automatically sent, unless you specify them as separate
  arguments.  Please see the application help for the new syntax.

* app_meetme: Mute and Unmute events are now reported via the Manager API.
  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
  are easier to use than "Action Command:". The MeetMeStopTalking event has
  also been deprecated in favor of the already existing MeetmeTalking event
  with a "Status" of "on" or "off" added.

* OriginateFailure and OriginateSuccess events were replaced by event
  OriginateResponse with a header named "Response" to indicate success or
  failure

Variables:

* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
  ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
  and ${LANGUAGE} have all been deprecated in favor of their related dialplan
  functions.  You are encouraged to move towards the associated dialplan
  function, as these variables will be removed in a future release.

* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now 
  adjustable from cdr.conf, instead of recompiling.

* OSP applications exports several new variables, ${OSPINHANDLE},
  ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
  ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
  
* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
  created channel. This variables holds the channel name of the transferer.

* The dial plan variable PRI_CAUSE will be removed from future versions 
  of Asterisk.
  It is replaced by adding a cause value to the hangup() application.

Functions:

* The function ${CHECK_MD5()} has been deprecated in favor of using an
  expression: $[${MD5(<string>)} = ${saved_md5}].

* The 'builtin' functions that used to be combined in pbx_functions.so are
  now built as separate modules. If you are not using 'autoload=yes' in your
  modules.conf file then you will need to explicitly load the modules that
  contain the functions you want to use.

* The ENUMLOOKUP() function with the 'c' option (for counting the number of 
  records), but the lookup fails to match any records, the returned value will 
  now be "0" instead of blank.

* The REALTIME() function is now available in version 1.4 and app_realtime has
  been deprecated in favor of the new function. app_realtime will be removed
  completely with the version 1.6 release so please take the time between
  releases to make any necessary changes

* The QUEUEAGENTCOUNT() function has been deprecated in favor of
  QUEUE_MEMBER_COUNT().

The IAX2 channel:

* It is possible that previous configurations depended on the order in which
  peers and users were specified in iax.conf for forcing the order in which
  chan_iax2 matched against them.  This behavior is going away and is considered
  deprecated in this version.  Avoid having ambiguous peer and user entries and
  to make things easy on yourself, always set the "username" option for users
  so that the remote end can match on that exactly instead of trying to infer
  which user you want based on host.

  If you would like to go ahead and use the new behavior which doesn't use the
  order in the config file to influence matching order, then change the 
  MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one.  An
  example is provided there.  By changing this, you will get *much* better
  performance on systems that do a lot of peer and user lookups as they will be
  stored in memory in a much more efficient manner.

* The "mailboxdetail" option has been deprecated.  Previously, if this option
  was not enabled, the 2 byte MSGCOUNT information element would be set to all
  1's to indicate there there is some number of messages waiting.  With this
  option enabled, the number of new messages were placed in one byte and the
  number of old messages are placed in the other.  This is now the default
  (and the only) behavior.

The SIP channel:

* The "incominglimit" setting is replaced by the "call-limit" setting in 
  sip.conf.

* OSP support code is removed from SIP channel to OSP applications. ospauth 
  option in sip.conf is removed to osp.conf as authpolicy. allowguest option
  in sip.conf cannot be set as osp anymore. 

* The Asterisk RTP stack has been changed in regards to RFC2833 reception
  and transmission. Packets will now be sent with proper duration instead of all
  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
  will want to turn on the rfc2833compensate option. Without this option your
  DTMF reception may act poorly.

* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
  in coming versions of Asterisk. Please use the dialplan function
  SIPCHANINFO(useragent) instead.

* The ALERT_INFO dialplan variable is deprecated and will be removed
  in coming versions of Asterisk. Please use the dialplan application
  sipaddheader() to add the "Alert-Info" header to the outbound invite.

* The "canreinvite" option has changed. canreinvite=yes used to disable
  re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
  to disable re-invites when NAT=yes. This is propably what you want.
  The settings are now: "yes", "no", "nonat", "update". Please consult
  sip.conf.sample for detailed information.

The Zap channel:

* Support for MFC/R2 has been removed, as it has not been functional for some
  time and it has no maintainer.

The Agent channel:

* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
  it provided can be done using dialplan logic, without requiring additional
  channel and module locks (which frequently caused deadlocks). An example of
  how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.

The G726-32 codec:

* It has been determined that previous versions of Asterisk used the wrong codeword
  packing order for G726-32 data. This version supports both available packing orders,
  and can transcode between them. It also now selects the proper order when
  negotiating with a SIP peer based on the codec name supplied in the SDP. However,
  there are existing devices that improperly request one order and then use another;
  Sipura and Grandstream ATAs are known to do this, and there may be others. To
  be able to continue to use these devices with this version of Asterisk and the
  G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
  to sip.conf, so that Asterisk can use the packing order expected by the device (even
  though it requested a different order). In addition, the internal format number for
  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
  result of this is that this version of Asterisk will be able to interoperate over
  IAX2 with older versions of Asterisk, as long as this version is told to allow
  'g726aal2' instead of 'g726' as the codec for the call.

Installation:

* On BSD systems, the installation directories have changed to more "FreeBSDish"
  directories. On startup, Asterisk will look for the main configuration in 
  /usr/local/etc/asterisk/asterisk.conf
  If you have an old installation, you might want to remove the binaries and 
  move the configuration files to the new locations. The following directories 
  are now default:
	ASTLIBDIR	/usr/local/lib/asterisk
	ASTVARLIBDIR	/usr/local/share/asterisk
	ASTETCDIR	/usr/local/etc/asterisk
	ASTBINDIR	/usr/local/bin/asterisk
	ASTSBINDIR	/usr/local/sbin/asterisk

Music on Hold:

* The music on hold handling has been changed in some significant ways in hopes
  to make it work in a way that is much less confusing to users. Behavior will
  not change if the same configuration is used from older versions of Asterisk.
  However, there are some new configuration options that will make things work
  in a way that makes more sense.

  Previously, many of the channel drivers had an option called "musicclass" or
  something similar. This option set what music on hold class this channel
  would *hear* when put on hold. Some people expected (with good reason) that
  this option was to configure what music on hold class to play when putting
  the bridged channel on hold. This option has now been deprecated.

  Two new music on hold related configuration options for channel drivers have
  been introduced. Some channel drivers support both options, some just one,
  and some support neither of them. Check the sample configuration files to see
  which options apply to which channel driver.

  The "mohsuggest" option specifies which music on hold class to suggest to the
  bridged channel when putting them on hold. The only way that this class can
  be overridden is if the bridged channel has a specific music class set that
  was done in the dialplan using Set(CHANNEL(musicclass)=something).

  The "mohinterpret" option is similar to the old "musicclass" option. It
  specifies which music on hold class this channel would like to listen to when
  put on hold. This music class is only effective if this channel has no music
  class set on it from the dialplan and the bridged channel putting this one on
  hold had no "mohsuggest" setting.

  The IAX2 and Zap channel drivers have an additional feature for the
  "mohinterpret" option. If this option is set to "passthrough", then these
  channel drivers will pass through the HOLD message in signalling instead of
  starting music on hold on the channel. An example for how this would be
  useful is in an enterprise network of Asterisk servers. When one phone on one
  server puts a phone on a different server on hold, the remote server will be
  responsible for playing the hold music to its local phone that was put on
  hold instead of the far end server across the network playing the music.

CDR Records:

* The behavior of the "clid" field of the CDR has always been that it will
  contain the callerid ANI if it is set, or the callerid number if ANI was not
  set.  When using the "callerid" option for various channel drivers, some
  would set ANI and some would not.  This has been cleared up so that all
  channel drivers set ANI.  If you would like to change the callerid number
  on the channel from the dialplan and have that change also show up in the 
  CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).

API:

* There are some API functions that were not previously prefixed with the 'ast_'
  prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
  have a module that uses the services provided by res_adsi, res_odbc, or
  res_agi, you will need to add ast_ prefixes to the functions that you call
  from those modules.

Formats:

* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
  in Asterisk 1.4.  This change was made in response to user complaints of
  choppiness or the clipping of loud signal peaks.  The GAIN preprocessor
  definition will be retained in Asterisk 1.4, but will be removed in a 
  future release.  The use of GAIN for the increasing of voicemail message
  volume should use the 'volgain' option in voicemail.conf

iLBC Codec:

* Previously, the Asterisk source code distribution included the iLBC
  encoder/decoder source code, from Global IP Solutions
  (http://www.gipscorp.com). This code is not licensed for
  distribution, and thus has been removed from the Asterisk source
  code distribution. If you wish to use codec_ilbc to support iLBC
  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
  script to download the source and put it in the proper place in
  the Asterisk build tree. Once that is done you can follow your normal
  steps of building Asterisk. You will need to run 'menuselect' and enable
  the iLBC codec in the 'Codec  Translators' category.