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At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well.

Browse Source 
This way, if people need to go back and review what was deprecated in previous
major releases, it is readily available to them.  Thanks for the suggestion!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Russell Bryant 18 years ago
parent ae6eea1999
commit 18347a73ff
3 changed files with 707 additions and 1 deletions
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211
UPGRADE-1.2.txt
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=========================================================
=== Information for upgrading from Asterisk 1.0 to 1.2
===
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
=========================================================
Compiling:
* The Asterisk 1.2 source code now uses C language features
supported only by 'modern' C compilers. Generally, this means GCC
version 3.0 or higher, although some GCC 2.96 releases will also
work. Some non-GCC compilers that support C99 and the common GCC
extensions (including anonymous structures and unions) will also
work. All releases of GCC 2.95 do _not_ have the requisite feature
support; systems using that compiler will need to be upgraded to
a more recent compiler release.
Dialplan Expressions:
* The dialplan expression parser (which handles $[ ... ] constructs)
has gone through a major upgrade, but has one incompatible change:
spaces are no longer required around expression operators, including
string comparisons. However, you can now use quoting to keep strings
together for comparison. For more details, please read the
doc/README.variables file, and check over your dialplan for possible
problems.
Agents:
* The default for ackcall has been changed to "no" instead of "yes"
because of a bug which caused the "yes" behavior to generally act like
"no". You may need to adjust the value if your agents behave
differently than you expect with respect to acknowledgement.
* The AgentCallBackLogin application now requires a second '|' before
specifying an extension@context. This is to distinguish the options
string from the extension, so that they do not conflict. See
'show application AgentCallbackLogin' for more details.
Parking:
* Parking behavior has changed slightly; when a parked call times out,
Asterisk will attempt to deliver the call back to the extension that
parked it, rather than the 's' extension. If that extension is busy
or unavailable, the parked call will be lost.
Dialing:
* The Caller*ID of the outbound leg is now the extension that was
called, rather than the Caller*ID of the inbound leg of the call. The
"o" flag for Dial can be used to restore the original behavior if
desired. Note that if you are looking for the originating callerid
from the manager event, there is a new manager event "Dial" which
provides the source and destination channels and callerid.
IAX:
* The naming convention for IAX channels has changed in two ways:
1. The call number follows a "-" rather than a "/" character.
2. The name of the channel has been simplified to IAX2/peer-callno,
rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
SIP:
* The global option "port" in 1.0.X that is used to set which port to
bind to has been changed to "bindport" to be more consistent with
the other channel drivers and to avoid confusion with the "port"
option for users/peers.
* The "Registry" event now uses "Username" rather than "User" for
consistency with IAX.
Applications:
* With the addition of dialplan functions (which operate similarly
to variables), the SetVar application has been renamed to Set.
* The CallerPres application has been removed. Use SetCallerPres
instead. It accepts both numeric and symbolic names.
* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
CheckGroup have been deprecated in favor of functions. Here is a
table of their replacements:
GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()})
GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test)
CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
Note that CheckGroup does not have a direct replacement. There is
also a new function called GROUP_LIST() which will return a space
separated list of all of the groups set on a channel. The GROUP()
function can also return the name of the group set on a channel when
used in a read environment.
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key) Set(foo=${DB(family/key)})
DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
* The application SetLanguage has been deprecated in favor of the
function LANGUAGE().
SetLanguage(fr) Set(LANGUAGE()=fr)
The LANGUAGE function can also return the currently set language:
Set(MYLANG=${LANGUAGE()})
* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
have been deprecated in favor of the function TIMEOUT(timeouttype):
AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300)
DigitTimeout(15) Set(TIMEOUT(digit)=15)
ResponseTimeout(15) Set(TIMEOUT(response)=15)
The TIMEOUT() function can also return the currently set timeouts:
Set(DTIMEOUT=${TIMEOUT(digit)})
* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
deprecated in favor of the CALLERID(datatype) function:
SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool)
SetCIDNum(2025551212) Set(CALLERID(number)=2025551212)
SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414)
* The application Record now uses the period to separate the filename
from the format, rather than the colon.
* The application VoiceMail now supports a 'temporary' greeting for each
mailbox. This greeting can be recorded by using option 4 in the
'mailbox options' menu, and 'change your password' option has been
moved to option 5.
* The application VoiceMailMain now only matches the 'default' context if
none is specified in the arguments. (This was the previously
documented behavior, however, we didn't follow that behavior.) The old
behavior can be restored by setting searchcontexts=yes in voicemail.conf.
Queues:
* A queue is now considered empty not only if there are no members but if
none of the members are available (e.g. agents not logged on). To
restore the original behavior, use "leavewhenempty=strict" or
"joinwhenempty=strict" instead of "=yes" for those options.
* It is now possible to use multi-digit extensions in the exit context
for a queue (although you should not have overlapping extensions,
as there is no digit timeout). This means that the EXITWITHKEY event
in queue_log can now contain a key field with more than a single
character in it.
Extensions:
* By default, there is a new option called "autofallthrough" in
extensions.conf that is set to yes. Asterisk 1.0 (and earlier)
behavior was to wait for an extension to be dialed after there were no
more extensions to execute. "autofallthrough" changes this behavior
so that the call will immediately be terminated with BUSY,
CONGESTION, or HANGUP based on Asterisk's best guess. If you are
writing an extension for IVR, you must use the WaitExten application
if "autofallthrough" is set to yes.
AGI:
* AGI scripts did not always get SIGHUP at the end, previously. That
behavior has been fixed. If you do not want your script to terminate
at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
be ignored within your application.
* CallerID is reported with agi_callerid and agi_calleridname instead
of a single parameter holding both.
Music On Hold:
* The preferred format for musiconhold.conf has changed; please see the
sample configuration file for the new format. The existing format
is still supported but will generate warnings when the module is loaded.
chan_modem:
* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
in this release, and will be removed in the next major Asterisk release.
Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
path for aopen and bestdata modem users.
MeetMe:
* The conference application now allows users to increase/decrease their
speaking volume and listening volume (independently of each other and
other users); the 'admin' and 'user' menus have changed, and new sound
files are included with this release. However, if a user calling in
over a Zaptel channel that does NOT have hardware DTMF detection
increases their speaking volume, it is likely they will no longer be
able to enter/exit the menu or make any further adjustments, as the
software DTMF detector will not be able to recognize the DTMF coming
from their device.
GetVar Manager Action:
* Previously, the behavior of the GetVar manager action reported the value
of a variable in the following manner:
> name: value
This has been changed to a manner similar to the SetVar action and is now
> Variable: name
> Value: value

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UPGRADE-1.4.txt
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=========================================================
=== Information for upgrading from Asterisk 1.2 to 1.4
===
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
=========================================================
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 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. 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

8
UPGRADE.txt
Unescape View File

@ -1,4 +1,10 @@
Information for Upgrading From Previous Asterisk Releases =========================================================
=== Information for upgrading from Asterisk 1.4 to 1.6
===
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
========================================================= =========================================================
AEL: AEL:

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