1 =========================================================
2 === Information for upgrading from Asterisk 1.0 to 1.2
5 === UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
6 === UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
7 === UPGRADE.txt -- Upgrade info for 1.4 to 1.6
8 =========================================================
12 * The Asterisk 1.2 source code now uses C language features
13 supported only by 'modern' C compilers. Generally, this means GCC
14 version 3.0 or higher, although some GCC 2.96 releases will also
15 work. Some non-GCC compilers that support C99 and the common GCC
16 extensions (including anonymous structures and unions) will also
17 work. All releases of GCC 2.95 do _not_ have the requisite feature
18 support; systems using that compiler will need to be upgraded to
19 a more recent compiler release.
23 * The dialplan expression parser (which handles $[ ... ] constructs)
24 has gone through a major upgrade, but has one incompatible change:
25 spaces are no longer required around expression operators, including
26 string comparisons. However, you can now use quoting to keep strings
27 together for comparison. For more details, please read the
28 doc/README.variables file, and check over your dialplan for possible
33 * The default for ackcall has been changed to "no" instead of "yes"
34 because of a bug which caused the "yes" behavior to generally act like
35 "no". You may need to adjust the value if your agents behave
36 differently than you expect with respect to acknowledgement.
38 * The AgentCallBackLogin application now requires a second '|' before
39 specifying an extension@context. This is to distinguish the options
40 string from the extension, so that they do not conflict. See
41 'show application AgentCallbackLogin' for more details.
45 * Parking behavior has changed slightly; when a parked call times out,
46 Asterisk will attempt to deliver the call back to the extension that
47 parked it, rather than the 's' extension. If that extension is busy
48 or unavailable, the parked call will be lost.
52 * The Caller*ID of the outbound leg is now the extension that was
53 called, rather than the Caller*ID of the inbound leg of the call. The
54 "o" flag for Dial can be used to restore the original behavior if
55 desired. Note that if you are looking for the originating callerid
56 from the manager event, there is a new manager event "Dial" which
57 provides the source and destination channels and callerid.
61 * The naming convention for IAX channels has changed in two ways:
62 1. The call number follows a "-" rather than a "/" character.
63 2. The name of the channel has been simplified to IAX2/peer-callno,
64 rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
68 * The global option "port" in 1.0.X that is used to set which port to
69 bind to has been changed to "bindport" to be more consistent with
70 the other channel drivers and to avoid confusion with the "port"
71 option for users/peers.
73 * The "Registry" event now uses "Username" rather than "User" for
78 * With the addition of dialplan functions (which operate similarly
79 to variables), the SetVar application has been renamed to Set.
81 * The CallerPres application has been removed. Use SetCallerPres
82 instead. It accepts both numeric and symbolic names.
84 * The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
85 CheckGroup have been deprecated in favor of functions. Here is a
86 table of their replacements:
88 GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()})
89 GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
90 SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test)
91 CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
93 Note that CheckGroup does not have a direct replacement. There is
94 also a new function called GROUP_LIST() which will return a space
95 separated list of all of the groups set on a channel. The GROUP()
96 function can also return the name of the group set on a channel when
97 used in a read environment.
99 * The applications DBGet and DBPut have been deprecated in favor of
100 functions. Here is a table of their replacements:
102 DBGet(foo=family/key) Set(foo=${DB(family/key)})
103 DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
105 * The application SetLanguage has been deprecated in favor of the
108 SetLanguage(fr) Set(LANGUAGE()=fr)
110 The LANGUAGE function can also return the currently set language:
112 Set(MYLANG=${LANGUAGE()})
114 * The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
115 have been deprecated in favor of the function TIMEOUT(timeouttype):
117 AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300)
118 DigitTimeout(15) Set(TIMEOUT(digit)=15)
119 ResponseTimeout(15) Set(TIMEOUT(response)=15)
121 The TIMEOUT() function can also return the currently set timeouts:
123 Set(DTIMEOUT=${TIMEOUT(digit)})
125 * The applications SetCIDName, SetCIDNum, and SetRDNIS have been
126 deprecated in favor of the CALLERID(datatype) function:
128 SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool)
129 SetCIDNum(2025551212) Set(CALLERID(number)=2025551212)
130 SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414)
132 * The application Record now uses the period to separate the filename
133 from the format, rather than the colon.
135 * The application VoiceMail now supports a 'temporary' greeting for each
136 mailbox. This greeting can be recorded by using option 4 in the
137 'mailbox options' menu, and 'change your password' option has been
140 * The application VoiceMailMain now only matches the 'default' context if
141 none is specified in the arguments. (This was the previously
142 documented behavior, however, we didn't follow that behavior.) The old
143 behavior can be restored by setting searchcontexts=yes in voicemail.conf.
147 * A queue is now considered empty not only if there are no members but if
148 none of the members are available (e.g. agents not logged on). To
149 restore the original behavior, use "leavewhenempty=strict" or
150 "joinwhenempty=strict" instead of "=yes" for those options.
152 * It is now possible to use multi-digit extensions in the exit context
153 for a queue (although you should not have overlapping extensions,
154 as there is no digit timeout). This means that the EXITWITHKEY event
155 in queue_log can now contain a key field with more than a single
160 * By default, there is a new option called "autofallthrough" in
161 extensions.conf that is set to yes. Asterisk 1.0 (and earlier)
162 behavior was to wait for an extension to be dialed after there were no
163 more extensions to execute. "autofallthrough" changes this behavior
164 so that the call will immediately be terminated with BUSY,
165 CONGESTION, or HANGUP based on Asterisk's best guess. If you are
166 writing an extension for IVR, you must use the WaitExten application
167 if "autofallthrough" is set to yes.
171 * AGI scripts did not always get SIGHUP at the end, previously. That
172 behavior has been fixed. If you do not want your script to terminate
173 at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
174 be ignored within your application.
176 * CallerID is reported with agi_callerid and agi_calleridname instead
177 of a single parameter holding both.
181 * The preferred format for musiconhold.conf has changed; please see the
182 sample configuration file for the new format. The existing format
183 is still supported but will generate warnings when the module is loaded.
187 * All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
188 in this release, and will be removed in the next major Asterisk release.
189 Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
190 path for aopen and bestdata modem users.
194 * The conference application now allows users to increase/decrease their
195 speaking volume and listening volume (independently of each other and
196 other users); the 'admin' and 'user' menus have changed, and new sound
197 files are included with this release. However, if a user calling in
198 over a Zaptel channel that does NOT have hardware DTMF detection
199 increases their speaking volume, it is likely they will no longer be
200 able to enter/exit the menu or make any further adjustments, as the
201 software DTMF detector will not be able to recognize the DTMF coming
204 GetVar Manager Action:
206 * Previously, the behavior of the GetVar manager action reported the value
207 of a variable in the following manner:
209 This has been changed to a manner similar to the SetVar action and is now