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1 =========================================================
2 === Information for upgrading from Asterisk 1.4 to 1.6
3 ===
4 ===
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 =========================================================
10 AEL:
12 * Macros are now implemented underneath with the Gosub() application.
13 Heaven Help You if you wrote code depending on any aspect of this!
14 Previous to 1.6, macros were implemented with the Macro() app, which
15 provided a nice feature of auto-returning. The compiler will do its
16 best to insert a Return() app call at the end of your macro if you did
17 not include it, but really, you should make sure that all execution
18 paths within your macros end in "return;".
20 * The conf2ael program is 'introduced' in this release; it is in a rather
21 crude state, but deemed useful for making a first pass at converting
22 extensions.conf code into AEL. More intelligence will come with time.
24 Core:
26 * The 'languageprefix' option in asterisk.conf is now deprecated, and
27 the default sound file layout for non-English sounds is the 'new
28 style' layout introduced in Asterisk 1.4 (and used by the automatic
29 sound file installer in the Makefile).
31 * The ast_expr2 stuff has been modified to handle floating-point numbers.
32 Numbers of the format D.D are now acceptable input for the expr parser,
33 Where D is a string of base-10 digits. All math is now done in "long double",
34 if it is available on your compiler/architecture. This was half-way between
35 a bug-fix (because the MATH func returns fp by default), and an enhancement.
36 Also, for those counting on, or needing, integer operations, a series of
37 'functions' were also added to the expr language, to allow several styles
38 of rounding/truncation, along with a set of common floating point operations,
39 like sin, cos, tan, log, pow, etc. The ability to call external functions
40 like CDR(), etc. was also added, without having to use the ${...} notation.
42 * The delimiter passed to applications has been changed to the comma (','), as
43 that is what people are used to using within extensions.conf. If you are
44 using realtime extensions, you will need to translate your existing dialplan
45 to use this separator. To use a literal comma, you need merely to escape it
46 with a backslash ('\'). Another possible side effect is that you may need to
47 remove the obscene level of backslashing that was necessary for the dialplan
48 to work correctly in 1.4 and previous versions. This should make writing
49 dialplans less painful in the future, albeit with the pain of a one-time
50 conversion.
52 * The logger.conf option 'rotatetimestamp' has been deprecated in favor of
53 'rotatestrategy'. This new option supports a 'rotate' strategy that more
54 closely mimics the system logger in terms of file rotation.
56 * The concise versions of various CLI commands are now deprecated. We recommend
57 using the manager interface (AMI) for application integration with Asterisk.
59 * The silencethreshold used for various applications is now settable via a
60 centralized config option in dsp.conf.
62 Voicemail:
64 * The voicemail configuration values 'maxmessage' and 'minmessage' have
65 been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
66 to make them more distinguishable from 'maxmsgs', which sets folder
67 size. The old variables will continue to work in this version, albeit
68 with a deprecation warning.
69 * If you use any interface for modifying voicemail aside from the built in
70 dialplan applications, then the option "pollmailboxes" *must* be set in
71 voicemail.conf for message waiting indication (MWI) to work properly. This
72 is because Voicemail notification is now event based instead of polling
73 based. The channel drivers are no longer responsible for constantly manually
74 checking mailboxes for changes so that they can send MWI information to users.
75 Examples of situations that would require this option are web interfaces to
76 voicemail or an email client in the case of using IMAP storage.
78 Applications:
80 * ChanIsAvail() now has a 't' option, which allows the specified device
81 to be queried for state without consulting the channel drivers. This
82 performs mostly a 'ChanExists' sort of function.
83 * ChannelRedirect() will not terminate the channel that fails to do a
84 channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
85 will reflect if the attempt was successful of not.
86 * SetCallerPres() has been replaced with the CALLERPRES() dialplan function
87 and is now deprecated.
88 * DISA()'s fifth argument is now an options argument. If you have previously
89 used 'NOANSWER' in this argument, you'll need to convert that to the new
90 option 'n'.
91 * Macro() is now deprecated. If you need subroutines, you should use the
92 Gosub()/Return() applications. To replace MacroExclusive(), we have
93 introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use
94 these functions in any location where you desire to ensure that only one
95 channel is executing that path at any one time.
96 * Read() now sets a READSTATUS variable on exit. It does NOT automatically
97 return -1 (and hangup) anymore on error. If you want to hangup on error,
98 you need to do so explicitly in your dialplan.
99 * Privacy() no longer uses privacy.conf, so any options must be specified
100 directly in the application arguments.
101 * MusicOnHold application now has duration parameter which allows specifying
102 timeout in seconds.
103 * WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
104 * SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
105 instead.
107 Dialplan Functions:
109 * QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
110 more information, issue a "show function QUEUE_MEMBER" from the CLI.
112 CDR:
114 * The cdr_sqlite module has been marked as deprecated in favor of
115 cdr_sqlite3_custom. It will potentially be removed from the tree
116 after Asterisk 1.6 is released.
118 * The cdr_odbc module now uses res_odbc to manage its connections. The
119 username and password parameters in cdr_odbc.conf, therefore, are no
120 longer used. The dsn parameter now points to an entry in res_odbc.conf.
122 Formats:
124 * format_wav: The GAIN preprocessor definition and source code that used it
125 is removed. This change was made in response to user complaints of
126 choppiness or the clipping of loud signal peaks. To increase the volume
127 of voicemail messages, use the 'volgain' option in voicemail.conf
129 Channel Drivers:
131 * SIP: a small upgrade to support the "Record" button on the SNOM360,
132 which sends a sip INFO message with a "Record: on" or "Record: off"
133 header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
134 requests (by default, via '*1'), then the user-configured dialpad sequence
135 is generated, and recording can be started and stopped via this button. The
136 file names and formats are all controlled via the normal mechanisms. If the
137 user has not configured the automon feature, the normal "415 Unsupported media type"
138 is returned, and nothing is done.
139 * SIP: The "call-limit" option is marked as deprecated. It still works in this version of
140 Asterisk, but will be removed in the following version. Please use the groupcount functions
141 in the dialplan to enforce call limits. The "limitonpeer" configuration option is
142 now renamed to "counteronpeer".
143 * SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
144 These are used only before registration to call a peer with the uri
145 sip:defaultuser@defaultip
146 The "username" setting still work, but is deprecated and will not work in
147 the next version of Asterisk.
149 * chan_local.c: the comma delimiter inside the channel name has been changed to a
150 semicolon, in order to make the Local channel driver compatible with the comma
151 delimiter change in applications.
152 * H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
153 to be compatible with settings in sip.conf. The "tos" and "cos" configuration
154 is deprecated and will stop working in the next release of Asterisk.
156 * Console: A new console channel driver, chan_console, has been added to Asterisk.
157 This new module can not be loaded at the same time as chan_alsa or chan_oss. The
158 default modules.conf only loads one of them (chan_oss by default). So, unless you
159 have modified your modules.conf to not use the autoload option, then you will need
160 to modify modules.conf to add another "noload" line to ensure that only one of
161 these three modules gets loaded.
163 * Zap: The "msdstrip" option has been deprecated, as it provides no value over
164 the method of stripping digits in the dialplan using variable substring syntax.
166 Configuration:
168 * pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
169 lowcost and other is not acceptable now. Look into qos.tex for description of
170 this parameter.
172 * queues.conf: the queue-lessthan sound file option is no longer available, and the
173 queue-round-seconds option no longer takes '1' as a valid parameter.
175 Manager:
177 * Manager has been upgraded to version 1.1 with a lot of changes.
178 Please check doc/manager_1_1.txt for information
180 * The IAXpeers command output has been changed to more closely resemble the
181 output of the SIPpeers command.
183 * cdr_manager now reports at the "cdr" level, not at "call" You may need to
184 change your manager.conf to add the level to existing AMI users, if they
185 want to see the CDR events generated.
187 * The Originate command now requires the Originate write permission. For
188 Originate with the Application parameter, you need the additional System
189 privilege if you want to do anything that calls out to a subshell.
191 iLBC Codec:
193 * Previously, the Asterisk source code distribution included the iLBC
194 encoder/decoder source code, from Global IP Solutions
195 (http://www.gipscorp.com). This code is not licensed for
196 distribution, and thus has been removed from the Asterisk source
197 code distribution. If you wish to use codec_ilbc to support iLBC
198 channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
199 script to download the source and put it in the proper place in
200 the Asterisk build tree. Once that is done you can follow your normal
201 steps of building Asterisk. You will need to run 'menuselect' and enable
202 the iLBC codec in the 'Codec Translators' category.