[VoIP] Asterisk 1.4

[Shane Young]

Quoting Jayson Smith <ratguy at bellsouth.net>:

> Hi again,
> I'm currently using Asterisk 1.2, and am wondering if I should upgrade. My
> current version is 1.2.3. What's new since my version, and especially in
> 1.4? If I were to upgrade, how much, especially in extensions.conf, is
> likely to break.

There are a few things:


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 FreePlay 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 FreePlay 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.

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.

* 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'.

* 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:

* 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.



--Shane
+1-821-7311 CNET