=== 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:
* A couple of CLI commands in res_ais were changed back to their original form:
"ais show clm members" --> "ais clm show members"
"ais show evt event channels" --> "ais evt show event channels"
* The default value for 'autofill' and 'shared_lastcall' in queues.conf has
been changed to 'yes'.
* The default value for the alwaysauthreject option in sip.conf has been changed
from "no" to "yes".
* 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
* 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
* 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:
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type)
so the dialplan can determine the B channel currently in use by the channel.
Use CHANNEL(no_media_path) to determine if the channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk
channel so AMI applications can passively determine the B channel currently
in use. Calls with "no-media" as the DAHDIChannel do not have an associated
B channel. No-media calls are either on hold or call-waiting.
* 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
* The 'stunaddr' option has been removed from chan_sip. This feature did not
behave as expected, had no correct use case, and was not RFC compliant. The
removal of this feature will hopefully be followed by a correct RFC compliant
STUN implementation in chan_sip in the future.
* The default value for the pedantic option in sip.conf has been changed
from "no" to "yes".
* The ConnectedLineNum and ConnectedLineName headers were added to many AMI
events/responses if the CallerIDNum/CallerIDName headers were also present.
The addition of connected line support changes the behavior of the channel
caller ID somewhat. The channel caller ID value no longer time shares with
the connected line ID on outgoing call legs. The timing of some AMI
events/responses output the connected line ID as caller ID. These party ID's
are now separate.
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
* 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
* 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 220.127.116.11 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
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'
* 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.