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 users,
18 that will result in a similar build to what they would have had before the
19 configure script was added to the build process (except for having to run 'make'
20 again after the configure script is run). Note that the configure script does NOT
21 need to be re-run just to rebuild Asterisk; you only need to re-run it when your
22 system configuration changes or you wish to build Asterisk with different options.
24 Build Process (module selection):
26 The Asterisk source tree now includes a basic module selection and build option
27 selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
28 In this tool, you can disable building of modules that you don't care about,
29 turn on/off global options for the build and see which modules will not (and cannot)
30 be built because your system does not have the required external dependencies
33 (TODO: document where 'global' and 'per-user' menuselect input files should go
34 and what they need to contain)
38 * The (very old and undocumented) ability to use BYEXTENSION for dialing
39 instead of ${EXTEN} has been removed.
41 Command Line Interface:
43 * 'show channels concise', designed to be used by applications that will parse
44 its output, previously used ':' characters to separate fields. However, some
45 of those fields can easily contain that character, making the output not
46 parseable. The delimiter has been changed to '!'.
50 * In previous Asterisk releases, many applications would jump to priority n+101
51 to indicate some kind of status or error condition. This functionality was
52 marked deprecated in Asterisk 1.2. An option to disable it was provided with
53 the default value set to 'on'. The default value for the global priority
54 jumping option is now 'off'.
56 * The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
57 AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
58 and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
59 been removed in this version. You should use the equivalent dialplan
60 function in places where you have previously used one of these applications.
62 * The application SetVar has been renamed to Set. The syntax SetVar was marked
63 deprecated in version 1.2 and is no longer recognized in this version.
65 * app_read has been updated to use the newer options codes, using "skip" or
66 "noanswer" will not work. Use s or n. Also there is a new feature i, for
67 using indication tones, so typing in skip would give you unexpected results.
69 * OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
73 * After executing the 'status' manager action, the "Status" manager events
74 included the header "CallerID:" which was actually only the CallerID number,
75 and not the full CallerID string. This header has been renamed to
76 "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
77 until after the release of 1.4, when it will be removed. Please use the time
78 during the 1.4 release to make this transition.
82 * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
83 ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
84 and ${LANGUAGE} have all been deprecated in favor of their related dialplan
85 functions. You are encouraged to move towards the associated dialplan
86 function, as these variables will be removed in a future release.
88 * The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
89 adjustable from cdr.conf, instead of recompiling.
91 * OSP applications exports several new variables, ${OSPINHANDLE},
92 ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
93 ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
97 * The function ${CHECK_MD5()} has been deprecated in favor of using an
98 expression: $[${MD5(<string>)} = ${saved_md5}].
100 * The 'builtin' functions that used to be combined in pbx_functions.so are
101 now built as separate modules. If you are not using 'autoload=yes' in your
102 modules.conf file then you will need to explicitly load the modules that
103 contain the functions you want to use.
105 * The ENUMLOOKUP() function with the 'c' option (for counting the number of records),
106 but the lookup fails to match any records, the returned value will now be "0" instead of blank.
110 * The "mailboxdetail" option has been deprecated. Previously, if this option
111 was not enabled, the 2 byte MSGCOUNT information element would be set to all
112 1's to indicate there there is some number of messages waiting. With this
113 option enabled, the number of new messages were placed in one byte and the
114 number of old messages are placed in the other. This is now the default
115 (and the only) behavior.
119 * The "incominglimit" setting is replaced by the "call-limit" setting in sip.conf.
121 * OSP support code is removed from SIP channel to OSP applications. ospauth
122 option in sip.conf is removed to osp.conf as authpolicy. allowguest option
123 in sip.conf cannot be set as osp anymore.
127 * On BSD systems, the installation directories have changed to more "FreeBSDish" directories. On startup, Asterisk will look for the main configuration in /usr/local/etc/asterisk/asterisk.conf
128 If you have an old installation, you might want to remove the binaries and move the configuration files to the new locations. The following directories are now default:
129 ASTLIBDIR /usr/local/lib/asterisk
130 ASTVARLIBDIR /usr/local/share/asterisk
131 ASTETCDIR /usr/local/etc/asterisk
132 ASTBINDIR /usr/local/bin/asterisk
133 ASTSBINDIR /usr/local/sbin/asterisk
137 * The phonetic sounds directory has been removed from the asterisk-sounds package
138 because they are now included directly in Asterisk. However, it is important to
139 note that the phonetic sounds that existed in asterisk-sounds used a different
140 naming convention than the sounds in Asterisk. For example, instead of alpha.gsm
141 and bravo.gsm, Asterisk has a_p.gsm and b_p.gsm.