![russell](/assets/img/avatar_default.png)
to all connected manager clients - update the UserEvent application to use the application argument parsing macros and to allow headers to be specified as pipe delimeted arguments (issue #5324, original patch by outtolunc, committed patch by Corydon) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@29017 f38db490-d61c-443f-a65b-d21fe96a405b
207 lines
10 KiB
Text
207 lines
10 KiB
Text
Information for Upgrading From Previous Asterisk Releases
|
|
=========================================================
|
|
|
|
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.
|
|
|
|
(TODO: document where 'global' and 'per-user' menuselect input files should go
|
|
and what they need to contain)
|
|
|
|
PBX Core:
|
|
|
|
* The (very old and undocumented) ability to use BYEXTENSION for dialing
|
|
instead of ${EXTEN} has been removed.
|
|
|
|
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 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.
|
|
|
|
* 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.
|
|
|
|
* 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.
|
|
|
|
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.
|
|
|
|
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}
|
|
|
|
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 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 Zap channel:
|
|
|
|
* Support for MFC/R2 has been removed, as it has not been functional for some time
|
|
and it has no maintainer.
|
|
|
|
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
|
|
|
|
Sounds:
|
|
|
|
* The phonetic sounds directory has been removed from the asterisk-sounds package
|
|
because they are now included directly in Asterisk. However, it is important to
|
|
note that the phonetic sounds that existed in asterisk-sounds used a different
|
|
naming convention than the sounds in Asterisk. For example, instead of alpha.gsm
|
|
and bravo.gsm, Asterisk has a_p.gsm and b_p.gsm.
|