490 lines
25 KiB
Text
490 lines
25 KiB
Text
![]() |
=========================================================
|
||
|
=== 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
|
||
|
|