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===

=== Information for upgrading from Asterisk 1.0 to 1.2

===

=== This file documents 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.

===

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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