mirror of https://github.com/asterisk/asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f310-digiumphones
Russell Bryant
15 years ago
parent
9a980d5acd
commit
c61b87c5f6
@ -0,0 +1,264 @@
|
||||
===========================================================
|
||||
===
|
||||
=== Information for upgrading between Asterisk versions
|
||||
===
|
||||
=== These files document all the changes that MUST be taken
|
||||
=== into account when upgrading between the Asterisk
|
||||
=== versions listed below. These changes may require that
|
||||
=== you modify your configuration files, dialplan or (in
|
||||
=== some cases) source code if you have your own Asterisk
|
||||
=== modules or patches. These files also includes advance
|
||||
=== notice of any functionality that has been marked as
|
||||
=== 'deprecated' and may be removed in a future release,
|
||||
=== along with the suggested replacement functionality.
|
||||
===
|
||||
=== 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-1.6.txt -- Upgrade info for 1.4 to 1.6
|
||||
===
|
||||
===========================================================
|
||||
|
||||
From 1.6.2 to 1.8:
|
||||
|
||||
* The behavior of the 'parkedcallstimeout' has changed slightly. The formulation
|
||||
of the extension name that a timed out parked call is delivered to when this
|
||||
option is set to 'no' was modified such that instead of converting '/' to '0',
|
||||
the '/' is converted to an underscore '_'. See the updated documentation in
|
||||
features.conf.sample for more information on the behavior of the
|
||||
'parkedcallstimeout' option.
|
||||
|
||||
* Asterisk-addons no longer exists as an independent package. Those modules
|
||||
now live in the addons directory of the main Asterisk source tree. They
|
||||
are not enabled by default. For more information about why modules live in
|
||||
addons, see README-addons.txt.
|
||||
|
||||
* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
|
||||
users of this channel in the tree have been converted to LOG_NOTICE or removed
|
||||
(in cases where the same message was already generated to another channel).
|
||||
|
||||
* The usage of RTP inside of Asterisk has now become modularized. This means
|
||||
the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
|
||||
If you are not using autoload=yes in modules.conf you will need to ensure
|
||||
it is set to load. If not, then any module which uses RTP (such as chan_sip)
|
||||
will not be able to send or receive calls.
|
||||
|
||||
* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
|
||||
remains. It now exists within app_chanspy.c and retains the exact same
|
||||
functionality as before.
|
||||
|
||||
* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
|
||||
1.6 behavior by default, if there is no [compat] section in asterisk.conf. In
|
||||
prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
|
||||
Specifically, that means that pbx_realtime and res_agi expect you to use commas
|
||||
to separate arguments in applications, and Set only takes a single pair of
|
||||
a variable name/value. The old 1.4 behavior may still be obtained by setting
|
||||
app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
|
||||
asterisk.conf.
|
||||
|
||||
* The PRI channels in chan_dahdi can no longer change the channel name if a
|
||||
different B channel is selected during call negotiation. To prevent using
|
||||
the channel name to infer what B channel a call is using and to avoid name
|
||||
collisions, the channel name format is changed.
|
||||
The new channel naming for PRI channels is:
|
||||
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
|
||||
|
||||
* The ChanIsAvail application has been changed so the AVAILSTATUS variable
|
||||
no longer contains both the device state and cause code. The cause code
|
||||
is now available in the AVAILCAUSECODE variable. If existing dialplan logic
|
||||
is written to expect AVAILSTATUS to contain the cause code it needs to be
|
||||
changed to use AVAILCAUSECODE.
|
||||
|
||||
* ExternalIVR will now send Z events for invalid or missing files, T events
|
||||
now include the interrupted file and bugs in argument parsing have been
|
||||
fixed so there may be arguments specified in incorrect ways that were
|
||||
working that will no longer work.
|
||||
Please see doc/externalivr.txt for details.
|
||||
|
||||
* OSP lookup application changes following variable names:
|
||||
OSPPEERIP to OSPINPEERIP
|
||||
OSPTECH to OSPOUTTECH
|
||||
OSPDEST to OSPDESTINATION
|
||||
OSPCALLING to OSPOUTCALLING
|
||||
OSPCALLED to OSPOUTCALLED
|
||||
OSPRESULTS to OSPDESTREMAILS
|
||||
|
||||
* The Manager event 'iax2 show peers' output has been updated. It now has a
|
||||
similar output of 'sip show peers'.
|
||||
|
||||
* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
|
||||
of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
|
||||
the current dialplan context.
|
||||
|
||||
* The CALLERPRES() dialplan function is deprecated in favor of
|
||||
CALLERID(num-pres) and CALLERID(name-pres).
|
||||
|
||||
* Environment variables that start with "AST_" are reserved to the system and
|
||||
may no longer be set from the dialplan.
|
||||
|
||||
* When a call is redirected inside of a Dial, the app and appdata fields of the
|
||||
CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
|
||||
|
||||
* The CDR handling of billsec and duration field has changed. If your table
|
||||
definition specifies those fields as float,double or similar they will now
|
||||
be logged with microsecond accuracy instead of a whole integer.
|
||||
|
||||
* chan_sip will no longer set up a local call forward when receiving a
|
||||
482 Loop Detected response. The dialplan will just continue from where it
|
||||
left off.
|
||||
|
||||
From 1.6.1 to 1.6.2:
|
||||
|
||||
* SIP no longer sends the 183 progress message for early media by
|
||||
default. Applications requiring early media should use the
|
||||
progress() dialplan app to generate the progress message.
|
||||
|
||||
* The firmware for the IAXy has been removed from Asterisk. It can be
|
||||
downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk
|
||||
install the firmware into its proper location, place the firmware in the
|
||||
contrib/firmware/iax/ directory in the Asterisk source tree before running
|
||||
"make install".
|
||||
|
||||
* T.38 FAX error correction mode can no longer be configured in udptl.conf;
|
||||
instead, it is configured on a per-peer (or global) basis in sip.conf, with
|
||||
the same default as was present in udptl.conf.sample.
|
||||
|
||||
* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
|
||||
instead, it is either supplied by the application servicing the T.38 channel
|
||||
(for a FAX send or receive) or calculated from the bridged endpoint's
|
||||
maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
|
||||
allows for overriding the value supplied by a remote endpoint, which is useful
|
||||
when T.38 connections are made to gateways that supply incorrectly-calculated
|
||||
maximum datagram sizes.
|
||||
|
||||
* There have been some changes to the IAX2 protocol to address the security
|
||||
concerns documented in the security advisory AST-2009-006. Please see the
|
||||
IAX2 security document, doc/IAX2-security.pdf, for information regarding
|
||||
backwards compatibility with versions of Asterisk that do not contain these
|
||||
changes to IAX2.
|
||||
|
||||
* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
|
||||
has been renamed to 'directmedia', to better reflect what it actually does.
|
||||
In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
|
||||
starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
|
||||
option never had any effect on these cases, it only affected the re-INVITEs
|
||||
used for direct media path setup. For MGCP and Skinny, the option was poorly
|
||||
named because those protocols don't even use INVITE messages at all. For
|
||||
backwards compatibility, the old option is still supported in both normal
|
||||
and Realtime configuration files, but all of the sample configuration files,
|
||||
Realtime/LDAP schemas, and other documentation refer to it using the new name.
|
||||
|
||||
* The default console now will use colors according to the default background
|
||||
color, instead of forcing the background color to black. If you are using a
|
||||
light colored background for your console, you may wish to use the option
|
||||
flag '-W' to present better color choices for the various messages. However,
|
||||
if you'd prefer the old method of forcing colors to white text on a black
|
||||
background, the compatibility option -B is provided for this purpose.
|
||||
|
||||
* SendImage() no longer hangs up the channel on transmission error or on
|
||||
any other error; in those cases, a FAILURE status is stored in
|
||||
SENDIMAGESTATUS and dialplan execution continues. The possible
|
||||
return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
|
||||
UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
|
||||
has been replaced with 'UNSUPPORTED'). This change makes the
|
||||
SendImage application more consistent with other applications.
|
||||
|
||||
* skinny.conf now has separate sections for lines and devices.
|
||||
Please have a look at configs/skinny.conf.sample and update
|
||||
your skinny.conf.
|
||||
|
||||
* Queue names previously were treated in a case-sensitive manner,
|
||||
meaning that queues with names like "sales" and "sALeS" would be
|
||||
seen as unique queues. The parsing logic has changed to use
|
||||
case-insensitive comparisons now when originally hashing based on
|
||||
queue names, meaning that now the two queues mentioned as examples
|
||||
earlier will be seen as having the same name.
|
||||
|
||||
* The SPRINTF() dialplan function has been moved into its own module,
|
||||
func_sprintf, and is no longer included in func_strings. If you use this
|
||||
function and do not use 'autoload=yes' in modules.conf, you will need
|
||||
to explicitly load func_sprintf for it to be available.
|
||||
|
||||
* The res_indications module has been removed. Its functionality was important
|
||||
enough that most of it has been moved into the Asterisk core.
|
||||
Two applications previously provided by res_indications, PlayTones and
|
||||
StopPlayTones, have been moved into a new module, app_playtones.
|
||||
|
||||
* Support for Taiwanese was incorrectly supported with the "tw" language code.
|
||||
In reality, the "tw" language code is reserved for the Twi language, native
|
||||
to Ghana. If you were previously using the "tw" language code, you should
|
||||
switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
|
||||
specific localizations. Additionally, "mx" should be changed to "es_MX",
|
||||
Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
|
||||
"cs", not "cz".
|
||||
|
||||
* DAHDISendCallreroutingFacility() parameters are now comma-separated,
|
||||
instead of the old pipe.
|
||||
|
||||
* res_jabber: autoprune has been disabled by default, to avoid misconfiguration
|
||||
that would end up being interpreted as a bug once Asterisk started removing
|
||||
the contacts from a user list.
|
||||
|
||||
* The cdr.conf file must exist and be configured correctly in order for CDR
|
||||
records to be written.
|
||||
|
||||
From 1.6.0.1 to 1.6.1:
|
||||
|
||||
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
|
||||
API calls were added in 1.6.0, so that modules that provide multiple
|
||||
AGI commands could register/unregister them all with a single
|
||||
step. However, these API calls were not implemented properly, and did
|
||||
not allow the caller to know whether registration or unregistration
|
||||
succeeded or failed. They have been redefined to now return success
|
||||
or failure, but this means any code using these functions will need
|
||||
be recompiled after upgrading to a version of Asterisk containing
|
||||
these changes. In addition, the source code using these functions
|
||||
should be reviewed to ensure it can properly react to failure
|
||||
of registration or unregistration of its API commands.
|
||||
|
||||
* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
|
||||
to better match what it really does, and the argument order has been
|
||||
changed to be consistent with other API calls that perform similar
|
||||
operations.
|
||||
|
||||
From 1.6.0.x to 1.6.1:
|
||||
|
||||
* In previous versions of Asterisk, due to the way objects were arranged in
|
||||
memory by chan_sip, the order of entries in sip.conf could be adjusted to
|
||||
control the behavior of matching against peers and users. The way objects
|
||||
are managed has been significantly changed for reasons involving performance
|
||||
and stability. A side effect of these changes is that the order of entries
|
||||
in sip.conf can no longer be relied upon to control behavior.
|
||||
|
||||
* The following core commands dealing with dialplan have been deprecated: 'core
|
||||
show globals', 'core set global' and 'core set chanvar'. Use the equivalent
|
||||
'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
|
||||
instead.
|
||||
|
||||
* In the dialplan expression parser, the logical value of spaces
|
||||
immediately preceding a standalone 0 previously evaluated to
|
||||
true. It now evaluates to false. This has confused a good many
|
||||
people in the past (typically because they failed to realize the
|
||||
space had any significance). Since this violates the Principle of
|
||||
Least Surprise, it has been changed.
|
||||
|
||||
* While app_directory has always relied on having a voicemail.conf or users.conf file
|
||||
correctly set up, it now is dependent on app_voicemail being compiled as well.
|
||||
|
||||
* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
|
||||
and you should start using that function instead for retrieving information about
|
||||
the channel in a technology-agnostic way.
|
||||
|
||||
* If you have any third party modules which use a config file variable whose
|
||||
name ends in a '+', please note that the append capability added to this
|
||||
version may now conflict with that variable naming scheme. An easy
|
||||
workaround is to ensure that a space occurs between the '+' and the '=',
|
||||
to differentiate your variable from the append operator. This potential
|
||||
conflict is unlikely, but is documented here to be thorough.
|
||||
|
||||
* The "Join" event from app_queue now uses the CallerIDNum header instead of
|
||||
the CallerID header to indicate the CallerID number.
|
||||
|
||||
* If you use ODBC storage for voicemail, there is a new field called "flag"
|
||||
which should be a char(8) or larger. This field specifies whether or not a
|
||||
message has been designated to be "Urgent", "PRIORITY", or not.
|
||||
|
||||
Loading…
Reference in new issue