1 Information for Upgrading From Previous Asterisk Releases
2 =========================================================
4 Build Process (configure script):
6 Asterisk now uses an autoconf-generated configuration script to learn how it
7 should build itself for your system. As it is a standard script, running:
11 will show you all the options available. This script can be used to tell the
12 build process what libraries you have on your system (if it cannot find them
13 automatically), which libraries you wish to have ignored even though they may
16 You must run the configure script before Asterisk will build, although it will
17 attempt to automatically run it for you with no options specified; for most
18 users, that will result in a similar build to what they would have had before
19 the configure script was added to the build process (except for having to run
20 'make' again after the configure script is run). Note that the configure script
21 does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
22 when your system configuration changes or you wish to build Asterisk with
25 Build Process (module selection):
27 The Asterisk source tree now includes a basic module selection and build option
28 selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
29 In this tool, you can disable building of modules that you don't care about,
30 turn on/off global options for the build and see which modules will not
31 (and cannot) be built because your system does not have the required external
32 dependencies installed.
34 The resulting file from menuselect is called 'menuselect.makeopts'. Note that
35 the resulting menuselect.makeopts file generally contains which modules *not*
36 to build. The modules listed in this file indicate which modules have unmet
37 dependencies, a present conflict, or have been disabled by the user in the
38 menuselect interface. Compiler Flags can also be set in the menuselect
39 interface. In this case, the resulting file contains which CFLAGS are in use,
40 not which ones are not in use.
42 If you would like to save your choices and have them applied against all
43 builds, the file can be copied to '~/.asterisk.makeopts' or
44 '/etc/asterisk.makeopts'.
48 * The (very old and undocumented) ability to use BYEXTENSION for dialing
49 instead of ${EXTEN} has been removed.
51 * Builtin (res_features) transfer functionality attempts to use the context
52 defined in TRANSFER_CONTEXT variable of the transferer channel first. If
53 not set, it uses the transferee variable. If not set in any channel, it will
54 attempt to use the last non macro context. If not possible, it will default
55 to the current context.
57 Command Line Interface:
59 * 'show channels concise', designed to be used by applications that will parse
60 its output, previously used ':' characters to separate fields. However, some
61 of those fields can easily contain that character, making the output not
62 parseable. The delimiter has been changed to '!'.
66 * In previous Asterisk releases, many applications would jump to priority n+101
67 to indicate some kind of status or error condition. This functionality was
68 marked deprecated in Asterisk 1.2. An option to disable it was provided with
69 the default value set to 'on'. The default value for the global priority
70 jumping option is now 'off'.
72 * The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
73 AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
74 and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
75 been removed in this version. You should use the equivalent dialplan
76 function in places where you have previously used one of these applications.
78 * The application SetGlobalVar has been deprecated. You should replace uses
79 of this application with the following combination of Set and GLOBAL():
80 Set(GLOBAL(name)=value). You may also access global variables exclusively by
81 using the GLOBAL() dialplan function, instead of relying on variable
82 interpolation falling back to globals when no channel variable is set.
84 * The application SetVar has been renamed to Set. The syntax SetVar was marked
85 deprecated in version 1.2 and is no longer recognized in this version.
87 * app_read has been updated to use the newer options codes, using "skip" or
88 "noanswer" will not work. Use s or n. Also there is a new feature i, for
89 using indication tones, so typing in skip would give you unexpected results.
91 * OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
93 * The CONNECT event in the queue_log from app_queue now has a second field
94 in addition to the holdtime field. It contains the unique ID of the
95 queue member channel that is taking the call. This is useful when trying
96 to link recording filenames back to a particular call from the queue.
98 * The old/current behavior of app_queue has a serial type behavior
99 in that the queue will make all waiting callers wait in the queue
100 even if there is more than one available member ready to take
101 calls until the head caller is connected with the member they
102 were trying to get to. The next waiting caller in line then
103 becomes the head caller, and they are then connected with the
104 next available member and all available members and waiting callers
105 waits while this happens. This cycle continues until there are
106 no more available members or waiting callers, whichever comes first.
107 The new behavior, enabled by setting autofill=yes in queues.conf
108 either at the [general] level to default for all queues or
109 to set on a per-queue level, makes sure that when the waiting
110 callers are connecting with available members in a parallel fashion
111 until there are no more available members or no more waiting callers,
112 whichever comes first. This is probably more along the lines of how
113 one would expect a queue should work and in most cases, you will want
114 to enable this new behavior. If you do not specify or comment out this
115 option, it will default to "no" to keep backward compatability with the old
118 * The app_queue application now has the ability to use MixMonitor to
119 record conversations queue members are having with queue callers. Please
120 see configs/queues.conf.sample for more information on this option.
122 * app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
123 the 'm' option now provides the functionality of "initially muted".
124 In practice, most existing dialplans using the 'm' flag should not notice
125 any difference, unless the keypad menu is enabled, allowing the user
126 to unmute themsleves.
128 * ast_play_and_record would attempt to cancel the recording if a DTMF
129 '0' was received. This behavior was not documented in most of the
130 applications that used ast_play_and_record and the return codes from
131 ast_play_and_record weren't checked for properly.
132 ast_play_and_record has been changed so that '0' no longer cancels a
133 recording. If you want to allow DTMF digits to cancel an
134 in-progress recording use ast_play_and_record_full which allows you
135 to specify which DTMF digits can be used to accept a recording and
136 which digits can be used to cancel a recording.
138 * ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
139 new ast_app_messagecount function which takes a single context/mailbox/folder
140 mailbox specification and returns the message count for that folder only.
141 This addresses the deficiency of not being able to count the number of
142 messages in folders other than INBOX and Old.
144 * The exit behavior of the AGI applications has changed. Previously, when
145 a connection to an AGI server failed, the application would cause the channel
146 to immediately stop dialplan execution and hangup. Now, the only time that
147 the AGI applications will cause the channel to stop dialplan execution is
148 when the channel itself requests hangup. The AGI applications now set an
149 AGISTATUS variable which will allow you to find out whether running the AGI
150 was successful or not.
152 Previously, there was no way to handle the case where Asterisk was unable to
153 locally execute an AGI script for some reason. In this case, dialplan
154 execution will continue as it did before, but the AGISTATUS variable will be
157 A locally executed AGI script can now exit with a non-zero exit code and this
158 failure will be detected by Asterisk. If an AGI script exits with a non-zero
159 exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
164 * After executing the 'status' manager action, the "Status" manager events
165 included the header "CallerID:" which was actually only the CallerID number,
166 and not the full CallerID string. This header has been renamed to
167 "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
168 until after the release of 1.4, when it will be removed. Please use the time
169 during the 1.4 release to make this transition.
171 * The AgentConnect event now has an additional field called "BridgedChannel"
172 which contains the unique ID of the queue member channel that is taking the
173 call. This is useful when trying to link recording filenames back to
174 a particular call from the queue.
176 * app_userevent has been modified to always send Event: UserEvent with the
177 additional header UserEvent: <userspec>. Also, the Channel and UniqueID
178 headers are not automatically sent, unless you specify them as separate
179 arguments. Please see the application help for the new syntax.
181 * app_meetme: Mute and Unmute events are now reported via the Manager API.
182 Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
183 are easier to use than "Action Command:". The MeetMeStopTalking event has
184 also been deprecated in favor of the already existing MeetmeTalking event
185 with a "Status" of "on" or "off" added.
189 * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
190 ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
191 and ${LANGUAGE} have all been deprecated in favor of their related dialplan
192 functions. You are encouraged to move towards the associated dialplan
193 function, as these variables will be removed in a future release.
195 * The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
196 adjustable from cdr.conf, instead of recompiling.
198 * OSP applications exports several new variables, ${OSPINHANDLE},
199 ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
200 ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
202 * Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
203 created channel. This variables holds the channel name of the transferer.
207 * The function ${CHECK_MD5()} has been deprecated in favor of using an
208 expression: $[${MD5(<string>)} = ${saved_md5}].
210 * The 'builtin' functions that used to be combined in pbx_functions.so are
211 now built as separate modules. If you are not using 'autoload=yes' in your
212 modules.conf file then you will need to explicitly load the modules that
213 contain the functions you want to use.
215 * The ENUMLOOKUP() function with the 'c' option (for counting the number of
216 records), but the lookup fails to match any records, the returned value will
217 now be "0" instead of blank.
219 * The REALTIME() function is now available in version 1.4 and app_realtime has
220 been deprecated in favor of the new function. app_realtime will be removed
221 completely with the version 1.6 release so please take the time between
222 releases to make any necessary changes
224 * The QUEUEAGENTCOUNT() function has been deprecated in favor of
225 QUEUE_MEMBER_COUNT().
229 * The "mailboxdetail" option has been deprecated. Previously, if this option
230 was not enabled, the 2 byte MSGCOUNT information element would be set to all
231 1's to indicate there there is some number of messages waiting. With this
232 option enabled, the number of new messages were placed in one byte and the
233 number of old messages are placed in the other. This is now the default
234 (and the only) behavior.
238 * The "incominglimit" setting is replaced by the "call-limit" setting in
241 * OSP support code is removed from SIP channel to OSP applications. ospauth
242 option in sip.conf is removed to osp.conf as authpolicy. allowguest option
243 in sip.conf cannot be set as osp anymore.
247 * Support for MFC/R2 has been removed, as it has not been functional for some
248 time and it has no maintainer.
252 * On BSD systems, the installation directories have changed to more "FreeBSDish"
253 directories. On startup, Asterisk will look for the main configuration in
254 /usr/local/etc/asterisk/asterisk.conf
255 If you have an old installation, you might want to remove the binaries and
256 move the configuration files to the new locations. The following directories
258 ASTLIBDIR /usr/local/lib/asterisk
259 ASTVARLIBDIR /usr/local/share/asterisk
260 ASTETCDIR /usr/local/etc/asterisk
261 ASTBINDIR /usr/local/bin/asterisk
262 ASTSBINDIR /usr/local/sbin/asterisk
266 * The phonetic sounds directory has been removed from the asterisk-sounds
267 package because they are now included directly in Asterisk. However, it is
268 important to note that the phonetic sounds that existed in asterisk-sounds
269 used a different naming convention than the sounds in Asterisk. For example,
270 instead of alpha.gsm and bravo.gsm, Asterisk has a_p.gsm and b_p.gsm.