mirror of https://github.com/asterisk/asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3changes/78/78/1
Matthew Jordan
13 years ago
parent
e2cd045c7e
commit
9d79ccd3db
@ -0,0 +1,226 @@
|
||||
===========================================================
|
||||
===
|
||||
=== 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 include 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
|
||||
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
|
||||
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
|
||||
===
|
||||
===========================================================
|
||||
|
||||
From 10 to 11:
|
||||
|
||||
Voicemail:
|
||||
- All voicemails now have a "msg_id" which uniquely identifies a message. For
|
||||
users of filesystem and IMAP storage of voicemail, this should be transparent.
|
||||
For users of ODBC, you will need to add a "msg_id" column to your voice mail
|
||||
messages table. This should be a string capable of holding at least 32 characters.
|
||||
All messages created in old Asterisk installations will have a msg_id added to
|
||||
them when required. This operation should be transparent as well.
|
||||
|
||||
Parking:
|
||||
- The comebacktoorigin setting must now be set per parking lot. The setting in
|
||||
the general section will not be applied automatically to each parking lot.
|
||||
- The BLINDTRANSFER channel variable is deleted from a channel when it is
|
||||
bridged to prevent subtle bugs in the parking feature. The channel
|
||||
variable is used by Asterisk internally for the Park application to work
|
||||
properly. If you were using it for your own purposes, copy it to your
|
||||
own channel variable before the channel is bridged.
|
||||
|
||||
res_ais:
|
||||
- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
|
||||
to use the res_corosync module, instead. OpenAIS is deprecated, but
|
||||
Corosync is still actively developed and maintained. Corosync came out of
|
||||
the OpenAIS project.
|
||||
|
||||
Dialplan Functions:
|
||||
- MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
|
||||
instead.
|
||||
- Macro has been deprecated in favor of GoSub. For redirecting and connected
|
||||
line purposes use the following variables instead of their macro equivalents:
|
||||
REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS,
|
||||
CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS.
|
||||
- The REDIRECTING function now supports the redirecting original party id
|
||||
and reason.
|
||||
- The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
|
||||
provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
|
||||
application has also been introduced to remove this data from the channel
|
||||
when necessary.
|
||||
|
||||
|
||||
func_enum:
|
||||
- ENUM query functions now return a count of -1 on lookup error to
|
||||
differentiate between a failed query and a successful query with 0 results
|
||||
matching the specified type.
|
||||
|
||||
CDR:
|
||||
- cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
|
||||
connect to databases that use schemas.
|
||||
|
||||
Configuration Files:
|
||||
- Files listed below have been updated to be more consistent with how Asterisk
|
||||
parses configuration files. This makes configuration files more consistent
|
||||
with what is expected across modules.
|
||||
|
||||
- cdr.conf: [general] and [csv] sections
|
||||
- dnsmgr.conf
|
||||
- dsp.conf
|
||||
|
||||
- The 'verbose' setting in logger.conf now takes an optional argument,
|
||||
specifying the verbosity level for each logging destination. The default,
|
||||
if not otherwise specified, is a verbosity of 3.
|
||||
|
||||
AMI:
|
||||
- DBDelTree now correctly returns an error when 0 rows are deleted just as
|
||||
the DBDel action does.
|
||||
- The IAX2 PeerStatus event now sends a 'Port' header. In Asterisk 10, this was
|
||||
erroneously being sent as a 'Post' header.
|
||||
|
||||
CCSS:
|
||||
- Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
|
||||
in channel configurations.
|
||||
|
||||
app_meetme:
|
||||
- The 'c' option (announce user count) will now work even if the 'q' (quiet)
|
||||
option is enabled.
|
||||
|
||||
app_followme:
|
||||
- Answered outgoing calls no longer get cut off when the next step is started.
|
||||
You now have until the last step times out to decide if you want to accept
|
||||
the call or not before being disconnected.
|
||||
|
||||
chan_gtalk:
|
||||
- chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
|
||||
that users switch to using it as it is a core supported module.
|
||||
|
||||
chan_jingle:
|
||||
- chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
|
||||
that users switch to using it as it is a core supported module.
|
||||
|
||||
SIP
|
||||
===
|
||||
- A new option "tonezone" for setting default tonezone for the channel driver
|
||||
or individual devices
|
||||
- A new manager event, "SessionTimeout" has been added and is triggered when
|
||||
a call is terminated due to RTP stream inactivity or SIP session timer
|
||||
expiration.
|
||||
- SIP_CAUSE is now deprecated. It has been modified to use the same
|
||||
mechanism as the HANGUPCAUSE function. Behavior should not change, but
|
||||
performance should be vastly improved. The HANGUPCAUSE function should now
|
||||
be used instead of SIP_CAUSE. Because of this, the storesipcause option in
|
||||
sip.conf is also deprecated.
|
||||
- The sip paramater for Originating Line Information (oli, isup-oli, and
|
||||
ss7-oli) is now parsed out of the From header and copied into the channel's
|
||||
ANI2 information field. This is readable from the CALLERID(ani2) dialplan
|
||||
function.
|
||||
- ICE support has been added and is enabled by default. Some endpoints may have
|
||||
problems with the ICE candidates within the SDP. If this is the case ICE support
|
||||
can be disabled globally or on a per-endpoint basis using the icesupport
|
||||
configuration option. Symptoms of this include one way media or no media flow.
|
||||
|
||||
chan_unistim
|
||||
- Due to massive update in chan_unistim phone keys functions and on-screen
|
||||
information changed.
|
||||
|
||||
users.conf:
|
||||
- A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
|
||||
as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
|
||||
documented in v1.4. Set the asterisk.conf stdexten=macro parameter to
|
||||
invoke the stdexten the old way.
|
||||
|
||||
res_jabber
|
||||
- This module has been deprecated in favor of the res_xmpp module. The res_xmpp
|
||||
module is backwards compatible with the res_jabber configuration file, dialplan
|
||||
functions, and AMI actions. The old CLI commands can also be made available using
|
||||
the res_clialiases template for Asterisk 11.
|
||||
|
||||
From 1.8 to 10:
|
||||
|
||||
cel_pgsql:
|
||||
- This module now expects an 'extra' column in the database for data added
|
||||
using the CELGenUserEvent() application.
|
||||
|
||||
ConfBridge
|
||||
- ConfBridge's dialplan arguments have changed and are not
|
||||
backwards compatible.
|
||||
|
||||
File Interpreters
|
||||
- The format interpreter formats/format_sln16.c for the file extension
|
||||
'.sln16' has been removed. The '.sln16' file interpreter now exists
|
||||
in the formats/format_sln.c module along with new support for sln12,
|
||||
sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
|
||||
|
||||
HTTP:
|
||||
- A bindaddr must be specified in order for the HTTP server
|
||||
to run. Previous versions would default to 0.0.0.0 if no
|
||||
bindaddr was specified.
|
||||
|
||||
Gtalk:
|
||||
- The default value for 'context' and 'parkinglots' in gtalk.conf has
|
||||
been changed to 'default', previously they were empty.
|
||||
|
||||
chan_dahdi:
|
||||
- The mohinterpret=passthrough setting is deprecated in favor of
|
||||
moh_signaling=notify.
|
||||
|
||||
pbx_lua:
|
||||
- Execution no longer continues after applications that do dialplan jumps
|
||||
(such as app.goto). Now when an application such as app.goto() is called,
|
||||
control is returned back to the pbx engine and the current extension
|
||||
function stops executing.
|
||||
- the autoservice now defaults to being on by default
|
||||
- autoservice_start() and autoservice_start() no longer return a value.
|
||||
|
||||
Queue:
|
||||
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
|
||||
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
|
||||
|
||||
Asterisk Database:
|
||||
- The internal Asterisk database has been switched from Berkeley DB 1.86 to
|
||||
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
|
||||
utility in the UTILS section of menuselect. If an existing astdb is found and no
|
||||
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
|
||||
convert an existing astdb to the SQLite3 version automatically at runtime. If
|
||||
moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
|
||||
to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
|
||||
|
||||
Manager:
|
||||
- The AMI protocol version was incremented to 1.2 as a result of changing two
|
||||
instances of the Unlink event to Bridge events. This change was documented
|
||||
as part of the AMI 1.1 update, but two Unlink events were inadvertently left
|
||||
unchanged.
|
||||
|
||||
Module Support Level
|
||||
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
|
||||
formats, funcs, pbx, and res have been updated to include MODULEINFO data
|
||||
that includes <support_level> tags with a value of core, extended, or deprecated.
|
||||
More information is available on the Asterisk wiki at
|
||||
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
|
||||
|
||||
Deprecated modules are now marked to not build by default and must be explicitly
|
||||
enabled in menuselect.
|
||||
|
||||
chan_sip:
|
||||
- Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
|
||||
by default. It can be enabled using the 'storesipcause' option. This feature
|
||||
has a significant performance penalty.
|
||||
|
||||
UDPTL:
|
||||
- The default UDPTL port range in udptl.conf.sample differed from the defaults
|
||||
in the source. If you didn't have a config file, you got 4500 to 4599. Now the
|
||||
default is 4000 to 4999.
|
||||
|
||||
===========================================================
|
||||
===========================================================
|
||||
Loading…
Reference in new issue