1 Information for Upgrading From Asterisk 1.0
2 ===========================================
6 * The Asterisk 1.2 source code now uses C language features
7 supported only by 'modern' C compilers. Generally, this means GCC
8 version 3.0 or higher, although some GCC 2.96 releases will also
9 work. Some non-GCC compilers that support C99 and the common GCC
10 extensions (including anonymous structures and unions) will also
11 work. All releases of GCC 2.95 do _not_ have the requisite feature
12 support; systems using that compiler will need to be upgraded to
13 a more recent compiler release.
17 * The dialplan expression parser (which handles $[ ... ] constructs)
18 has gone through a major upgrade, but has one incompatible change:
19 spaces are no longer required around expression operators, including
20 string comparisons. However, you can now use quoting to keep strings
21 together for comparison. For more details, please read the
22 doc/README.variables file, and check over your dialplan for possible
27 * The default for ackcall has been changed to "no" instead of "yes"
28 because of a bug which caused the "yes" behavior to generally act like
29 "no". You may need to adjust the value if your agents behave
30 differently than you expect with respect to acknowledgement.
32 * The AgentCallBackLogin application now requires a second '|' before
33 specifying an extension@context. This is to distinguish the options
34 string from the extension, so that they do not conflict. See
35 'show application AgentCallbackLogin' for more details.
39 * The Caller*ID of the outbound leg is now the extension that was
40 called, rather than the Caller*ID of the inbound leg of the call. The
41 "o" flag for Dial can be used to restore the original behavior if
42 desired. Note that if you are looking for the originating callerid
43 from the manager event, there is a new manager event "Dial" which
44 provides the source and destination channels and callerid.
48 * The naming convention for IAX channels has changed in a minor way such
49 that the call number follows a "-" rather than a "/" character.
53 * The global option "port" in 1.0.X that is used to set which port to
54 bind to has been changed to "bindport" to be more consistent with
55 the other channel drivers and to avoid confusion with the "port"
56 option for users/peers.
58 * The "Registry" event now uses "Username" rather than "User" for
63 * With the addition of dialplan functions (which operate similarly
64 to variables), the SetVar application has been renamed to Set.
66 * The CallerPres application has been removed. Use SetCallerPres
67 instead. It accepts both numeric and symbolic names.
69 * The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
70 CheckGroup have been deprecated in favor of functions. Here is a
71 table of their replacements:
73 GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()})
74 GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
75 SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test)
76 CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
78 Note that CheckGroup does not have a direct replacement. There is
79 also a new function called GROUP_LIST() which will return a space
80 separated list of all of the groups set on a channel. The GROUP()
81 function can also return the name of the group set on a channel when
82 used in a read environment.
84 * The applications DBGet and DBPut have been deprecated in favor of
85 functions. Here is a table of their replacements:
87 DBGet(foo=family/key) Set(foo=${DB(family/key)})
88 DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
90 * The application SetLanguage has been deprecated in favor of the
93 SetLanguage(fr) Set(LANGUAGE()=fr)
95 The LANGUAGE function can also return the currently set language:
97 Set(MYLANG=${LANGUAGE()})
99 * The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
100 have been deprecated in favor of the function TIMEOUT(timeouttype):
102 AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300)
103 DigitTimeout(15) Set(TIMEOUT(digit)=15)
104 ResponseTimeout(15) Set(TIMEOUT(response)=15)
106 The TIMEOUT() function can also return the currently set timeouts:
108 Set(DTIMEOUT=${TIMEOUT(digit)})
110 * The applications SetCIDName, SetCIDNum, and SetRDNIS have been
111 deprecated in favor of the CALLERID(datatype) function:
113 SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool)
114 SetCIDNum(2025551212) Set(CALLERID(number)=2025551212)
115 SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414)
117 * The application Record now uses the period to separate the filename
118 from the format, rather than the colon.
120 * The application VoiceMail now supports a 'temporary' greeting for each
121 mailbox. This greeting can be recorded by using option 4 in the
122 'mailbox options' menu, and 'change your password' option has been
127 * A queue is now considered empty not only if there are no members but if
128 none of the members are available (e.g. agents not logged on). To
129 restore the original behavior, use "leavewhenempty=strict" or
130 "joinwhenempty=strict" instead of "=yes" for those options.
132 * It is now possible to use multi-digit extensions in the exit context
133 for a queue (although you should not have overlapping extensions,
134 as there is no digit timeout). This means that the EXITWITHKEY event
135 in queue_log can now contain a key field with more than a single
140 * By default, there is a new option called "autofallthrough" in
141 extensions.conf that is set to yes. Asterisk 1.0 (and earlier)
142 behavior was to wait for an extension to be dialed after there were no
143 more extensions to execute. "autofallthrough" changes this behavior
144 so that the call will immediately be terminated with BUSY,
145 CONGESTION, or HANGUP based on Asterisk's best guess. If you are
146 writing an extension for IVR, you must use the WaitExten application
147 if "autofallthrough" is set to yes.
151 * AGI scripts did not always get SIGHUP at the end, previously. That
152 behavior has been fixed. If you do not want your script to terminate
153 at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
154 be ignored within your application.
158 * The preferred format for musiconhold.conf has changed; please see the
159 sample configuration file for the new format. The existing format
160 is still supported but will generate warnings when the module is loaded.
164 * All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
165 in this release, and will be removed in the next major Asterisk release.
166 Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
167 path for aopen and bestdata modem users.
171 * The conference application now allows users to increase/decrease their
172 speaking volume and listening volume (independently of each other and
173 other users); the 'admin' and 'user' menus have changed, and new sound
174 files are included with this release. However, if a user calling in
175 over a Zaptel channel that does NOT have hardware DTMF detection
176 increases their speaking volume, it is likely they will no longer be
177 able to enter/exit the menu or make any further adjustments, as the
178 software DTMF detector will not be able to recognize the DTMF coming
181 GetVar Manager Action:
183 * Previously, the behavior of the GetVar manager action reported the value
184 of a variable in the following manner:
186 This has been changed to a manner similar to the SetVar action and is now