1 =========================================================
3 === Information for upgrading from Asterisk 1.2 to 1.4
5 === These files document all the changes that MUST be taken
6 === into account when upgrading between the Asterisk
7 === versions listed below. These changes may require that
8 === you modify your configuration files, dialplan or (in
9 === some cases) source code if you have your own Asterisk
10 === modules or patches. These files also includes advance
11 === notice of any functionality that has been marked as
12 === 'deprecated' and may be removed in a future release,
13 === along with the suggested replacement functionality.
15 === UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
17 =========================================================
19 Build Process (configure script):
21 Asterisk now uses an autoconf-generated configuration script to learn how it
22 should build itself for your system. As it is a standard script, running:
26 will show you all the options available. This script can be used to tell the
27 build process what libraries you have on your system (if it cannot find them
28 automatically), which libraries you wish to have ignored even though they may
31 You must run the configure script before Asterisk will build, although it will
32 attempt to automatically run it for you with no options specified; for most
33 users, that will result in a similar build to what they would have had before
34 the configure script was added to the build process (except for having to run
35 'make' again after the configure script is run). Note that the configure script
36 does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
37 when your system configuration changes or you wish to build Asterisk with
40 Build Process (module selection):
42 The Asterisk source tree now includes a basic module selection and build option
43 selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
44 In this tool, you can disable building of modules that you don't care about,
45 turn on/off global options for the build and see which modules will not
46 (and cannot) be built because your system does not have the required external
47 dependencies installed.
49 The resulting file from menuselect is called 'menuselect.makeopts'. Note that
50 the resulting menuselect.makeopts file generally contains which modules *not*
51 to build. The modules listed in this file indicate which modules have unmet
52 dependencies, a present conflict, or have been disabled by the user in the
53 menuselect interface. Compiler Flags can also be set in the menuselect
54 interface. In this case, the resulting file contains which CFLAGS are in use,
55 not which ones are not in use.
57 If you would like to save your choices and have them applied against all
58 builds, the file can be copied to '~/.asterisk.makeopts' or
59 '/etc/asterisk.makeopts'.
61 Build Process (Makefile targets):
63 The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
64 is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
65 in the menuselect tool.
67 It is now possible to run most make targets against a single subdirectory; from
68 the top level directory, for example, 'make channels' will run 'make all' in the
69 'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
71 Sound (prompt) and Music On Hold files:
73 Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
74 use with Asterisk have been replaced with new versions produced from high quality
75 master recordings, and are available in three languages (English, French and
76 Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
77 In addition, the music on hold files provided by opsound.org Music are now available
78 in the same five formats, but no longer available in MP3 format.
80 The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
81 (as were supplied with previous releases) and the opsound.org MOH files in WAV format.
82 All of the other variations can be installed by running 'make menuselect' and
83 selecting the packages you wish to install; when you run 'make install', those
84 packages will be downloaded and installed along with the standard files included
87 If for some reason you expect to not have Internet access at the time you will be
88 running 'make install', you can make your package selections using menuselect and
89 then run 'make sounds' to download (only) the sound packages; this will leave the
90 sound packages in the 'sounds' subdirectory to be used later during installation.
92 WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
93 instead of the alternate-language files being stored in subdirectories underneath
94 the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
95 etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
96 language itself, then places all the sound files for that language under that
97 directory and its subdirectories. This is the layout that will be created if you
98 select non-English languages to be installed via menuselect, HOWEVER Asterisk does
99 not default to this layout and will not find the files in the places it expects them
100 to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
101 /etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
106 * The (very old and undocumented) ability to use BYEXTENSION for dialing
107 instead of ${EXTEN} has been removed.
109 * Builtin (res_features) transfer functionality attempts to use the context
110 defined in TRANSFER_CONTEXT variable of the transferer channel first. If
111 not set, it uses the transferee variable. If not set in any channel, it will
112 attempt to use the last non macro context. If not possible, it will default
113 to the current context.
115 * The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
116 if your dialplan relies on the ability to 'run off the end' of an extension
117 and wait for a new extension without using WaitExten() to accomplish that,
118 you will need set autofallthrough to 'no' in your extensions.conf file.
120 Command Line Interface:
122 * 'show channels concise', designed to be used by applications that will parse
123 its output, previously used ':' characters to separate fields. However, some
124 of those fields can easily contain that character, making the output not
125 parseable. The delimiter has been changed to '!'.
129 * In previous Asterisk releases, many applications would jump to priority n+101
130 to indicate some kind of status or error condition. This functionality was
131 marked deprecated in Asterisk 1.2. An option to disable it was provided with
132 the default value set to 'on'. The default value for the global priority
133 jumping option is now 'off'.
135 * The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
136 AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
137 and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
138 been removed in this version. You should use the equivalent dialplan
139 function in places where you have previously used one of these applications.
141 * The application SetGlobalVar has been deprecated. You should replace uses
142 of this application with the following combination of Set and GLOBAL():
143 Set(GLOBAL(name)=value). You may also access global variables exclusively by
144 using the GLOBAL() dialplan function, instead of relying on variable
145 interpolation falling back to globals when no channel variable is set.
147 * The application SetVar has been renamed to Set. The syntax SetVar was marked
148 deprecated in version 1.2 and is no longer recognized in this version. The
149 use of Set with multiple argument pairs has also been deprecated. Please
150 separate each name/value pair into its own dialplan line.
152 * app_read has been updated to use the newer options codes, using "skip" or
153 "noanswer" will not work. Use s or n. Also there is a new feature i, for
154 using indication tones, so typing in skip would give you unexpected results.
156 * OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
158 * The CONNECT event in the queue_log from app_queue now has a second field
159 in addition to the holdtime field. It contains the unique ID of the
160 queue member channel that is taking the call. This is useful when trying
161 to link recording filenames back to a particular call from the queue.
163 * The old/current behavior of app_queue has a serial type behavior
164 in that the queue will make all waiting callers wait in the queue
165 even if there is more than one available member ready to take
166 calls until the head caller is connected with the member they
167 were trying to get to. The next waiting caller in line then
168 becomes the head caller, and they are then connected with the
169 next available member and all available members and waiting callers
170 waits while this happens. This cycle continues until there are
171 no more available members or waiting callers, whichever comes first.
172 The new behavior, enabled by setting autofill=yes in queues.conf
173 either at the [general] level to default for all queues or
174 to set on a per-queue level, makes sure that when the waiting
175 callers are connecting with available members in a parallel fashion
176 until there are no more available members or no more waiting callers,
177 whichever comes first. This is probably more along the lines of how
178 one would expect a queue should work and in most cases, you will want
179 to enable this new behavior. If you do not specify or comment out this
180 option, it will default to "no" to keep backward compatability with the old
183 * Queues depend on the channel driver reporting the proper state
184 for each member of the queue. To get proper signalling on
185 queue members that use the SIP channel driver, you need to
186 enable a call limit (could be set to a high value so it
187 is not put into action) and also make sure that both inbound
188 and outbound calls are accounted for.
200 * The app_queue application now has the ability to use MixMonitor to
201 record conversations queue members are having with queue callers. Please
202 see configs/queues.conf.sample for more information on this option.
204 * The app_queue application strategy called 'roundrobin' has been deprecated
205 for this release. Users are encouraged to use 'rrmemory' instead, since it
206 provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
207 'rrmemory' will be renamed 'roundrobin'.
209 * The app_queue application option called 'monitor-join' has been deprecated
210 for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
211 since it provides the same functionality but is not dependent on soxmix or some
212 other external program in order to mix the audio.
214 * app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
215 the 'm' option now provides the functionality of "initially muted".
216 In practice, most existing dialplans using the 'm' flag should not notice
217 any difference, unless the keypad menu is enabled, allowing the user
218 to unmute themsleves.
220 * ast_play_and_record would attempt to cancel the recording if a DTMF
221 '0' was received. This behavior was not documented in most of the
222 applications that used ast_play_and_record and the return codes from
223 ast_play_and_record weren't checked for properly.
224 ast_play_and_record has been changed so that '0' no longer cancels a
225 recording. If you want to allow DTMF digits to cancel an
226 in-progress recording use ast_play_and_record_full which allows you
227 to specify which DTMF digits can be used to accept a recording and
228 which digits can be used to cancel a recording.
230 * ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
231 new ast_app_messagecount function which takes a single context/mailbox/folder
232 mailbox specification and returns the message count for that folder only.
233 This addresses the deficiency of not being able to count the number of
234 messages in folders other than INBOX and Old.
236 * The exit behavior of the AGI applications has changed. Previously, when
237 a connection to an AGI server failed, the application would cause the channel
238 to immediately stop dialplan execution and hangup. Now, the only time that
239 the AGI applications will cause the channel to stop dialplan execution is
240 when the channel itself requests hangup. The AGI applications now set an
241 AGISTATUS variable which will allow you to find out whether running the AGI
242 was successful or not.
244 Previously, there was no way to handle the case where Asterisk was unable to
245 locally execute an AGI script for some reason. In this case, dialplan
246 execution will continue as it did before, but the AGISTATUS variable will be
249 A locally executed AGI script can now exit with a non-zero exit code and this
250 failure will be detected by Asterisk. If an AGI script exits with a non-zero
251 exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
254 * app_voicemail: The ODBC_STORAGE capability now requires the extended table format
255 previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
256 your table format using the schema provided in doc/odbcstorage.txt
258 * app_waitforsilence: Fixes have been made to this application which changes the
259 default behavior with how quickly it returns. You can maintain "old-style" behavior
260 with the addition/use of a third "timeout" parameter.
261 Please consult the application documentation and make changes to your dialplan
266 * After executing the 'status' manager action, the "Status" manager events
267 included the header "CallerID:" which was actually only the CallerID number,
268 and not the full CallerID string. This header has been renamed to
269 "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
270 until after the release of 1.4, when it will be removed. Please use the time
271 during the 1.4 release to make this transition.
273 * The AgentConnect event now has an additional field called "BridgedChannel"
274 which contains the unique ID of the queue member channel that is taking the
275 call. This is useful when trying to link recording filenames back to
276 a particular call from the queue.
278 * app_userevent has been modified to always send Event: UserEvent with the
279 additional header UserEvent: <userspec>. Also, the Channel and UniqueID
280 headers are not automatically sent, unless you specify them as separate
281 arguments. Please see the application help for the new syntax.
283 * app_meetme: Mute and Unmute events are now reported via the Manager API.
284 Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
285 are easier to use than "Action Command:". The MeetMeStopTalking event has
286 also been deprecated in favor of the already existing MeetmeTalking event
287 with a "Status" of "on" or "off" added.
289 * OriginateFailure and OriginateSuccess events were replaced by event
290 OriginateResponse with a header named "Response" to indicate success or
295 * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
296 ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
297 and ${LANGUAGE} have all been deprecated in favor of their related dialplan
298 functions. You are encouraged to move towards the associated dialplan
299 function, as these variables will be removed in a future release.
301 * The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
302 adjustable from cdr.conf, instead of recompiling.
304 * OSP applications exports several new variables, ${OSPINHANDLE},
305 ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
306 ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
308 * Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
309 created channel. This variables holds the channel name of the transferer.
311 * The dial plan variable PRI_CAUSE will be removed from future versions
313 It is replaced by adding a cause value to the hangup() application.
317 * The function ${CHECK_MD5()} has been deprecated in favor of using an
318 expression: $[${MD5(<string>)} = ${saved_md5}].
320 * The 'builtin' functions that used to be combined in pbx_functions.so are
321 now built as separate modules. If you are not using 'autoload=yes' in your
322 modules.conf file then you will need to explicitly load the modules that
323 contain the functions you want to use.
325 * The ENUMLOOKUP() function with the 'c' option (for counting the number of
326 records), but the lookup fails to match any records, the returned value will
327 now be "0" instead of blank.
329 * The REALTIME() function is now available in version 1.4 and app_realtime has
330 been deprecated in favor of the new function. app_realtime will be removed
331 completely with the version 1.6 release so please take the time between
332 releases to make any necessary changes
334 * The QUEUEAGENTCOUNT() function has been deprecated in favor of
335 QUEUE_MEMBER_COUNT().
339 * It is possible that previous configurations depended on the order in which
340 peers and users were specified in iax.conf for forcing the order in which
341 chan_iax2 matched against them. This behavior is going away and is considered
342 deprecated in this version. Avoid having ambiguous peer and user entries and
343 to make things easy on yourself, always set the "username" option for users
344 so that the remote end can match on that exactly instead of trying to infer
345 which user you want based on host.
347 If you would like to go ahead and use the new behavior which doesn't use the
348 order in the config file to influence matching order, then change the
349 MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one. An
350 example is provided there. By changing this, you will get *much* better
351 performance on systems that do a lot of peer and user lookups as they will be
352 stored in memory in a much more efficient manner.
354 * The "mailboxdetail" option has been deprecated. Previously, if this option
355 was not enabled, the 2 byte MSGCOUNT information element would be set to all
356 1's to indicate there there is some number of messages waiting. With this
357 option enabled, the number of new messages were placed in one byte and the
358 number of old messages are placed in the other. This is now the default
359 (and the only) behavior.
363 * The "incominglimit" setting is replaced by the "call-limit" setting in
366 * OSP support code is removed from SIP channel to OSP applications. ospauth
367 option in sip.conf is removed to osp.conf as authpolicy. allowguest option
368 in sip.conf cannot be set as osp anymore.
370 * The Asterisk RTP stack has been changed in regards to RFC2833 reception
371 and transmission. Packets will now be sent with proper duration instead of all
372 at once. If you are receiving calls from a pre-1.4 Asterisk installation you
373 will want to turn on the rfc2833compensate option. Without this option your
374 DTMF reception may act poorly.
376 * The $SIPUSERAGENT dialplan variable is deprecated and will be removed
377 in coming versions of Asterisk. Please use the dialplan function
378 SIPCHANINFO(useragent) instead.
380 * The ALERT_INFO dialplan variable is deprecated and will be removed
381 in coming versions of Asterisk. Please use the dialplan application
382 sipaddheader() to add the "Alert-Info" header to the outbound invite.
384 * The "canreinvite" option has changed. canreinvite=yes used to disable
385 re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
386 to disable re-invites when NAT=yes. This is propably what you want.
387 The settings are now: "yes", "no", "nonat", "update". Please consult
388 sip.conf.sample for detailed information.
392 * Support for MFC/R2 has been removed, as it has not been functional for some
393 time and it has no maintainer.
397 * Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
398 it provided can be done using dialplan logic, without requiring additional
399 channel and module locks (which frequently caused deadlocks). An example of
400 how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
404 * It has been determined that previous versions of Asterisk used the wrong codeword
405 packing order for G726-32 data. This version supports both available packing orders,
406 and can transcode between them. It also now selects the proper order when
407 negotiating with a SIP peer based on the codec name supplied in the SDP. However,
408 there are existing devices that improperly request one order and then use another;
409 Sipura and Grandstream ATAs are known to do this, and there may be others. To
410 be able to continue to use these devices with this version of Asterisk and the
411 G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
412 to sip.conf, so that Asterisk can use the packing order expected by the device (even
413 though it requested a different order). In addition, the internal format number for
414 G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
415 result of this is that this version of Asterisk will be able to interoperate over
416 IAX2 with older versions of Asterisk, as long as this version is told to allow
417 'g726aal2' instead of 'g726' as the codec for the call.
421 * On BSD systems, the installation directories have changed to more "FreeBSDish"
422 directories. On startup, Asterisk will look for the main configuration in
423 /usr/local/etc/asterisk/asterisk.conf
424 If you have an old installation, you might want to remove the binaries and
425 move the configuration files to the new locations. The following directories
427 ASTLIBDIR /usr/local/lib/asterisk
428 ASTVARLIBDIR /usr/local/share/asterisk
429 ASTETCDIR /usr/local/etc/asterisk
430 ASTBINDIR /usr/local/bin/asterisk
431 ASTSBINDIR /usr/local/sbin/asterisk
435 * The music on hold handling has been changed in some significant ways in hopes
436 to make it work in a way that is much less confusing to users. Behavior will
437 not change if the same configuration is used from older versions of Asterisk.
438 However, there are some new configuration options that will make things work
439 in a way that makes more sense.
441 Previously, many of the channel drivers had an option called "musicclass" or
442 something similar. This option set what music on hold class this channel
443 would *hear* when put on hold. Some people expected (with good reason) that
444 this option was to configure what music on hold class to play when putting
445 the bridged channel on hold. This option has now been deprecated.
447 Two new music on hold related configuration options for channel drivers have
448 been introduced. Some channel drivers support both options, some just one,
449 and some support neither of them. Check the sample configuration files to see
450 which options apply to which channel driver.
452 The "mohsuggest" option specifies which music on hold class to suggest to the
453 bridged channel when putting them on hold. The only way that this class can
454 be overridden is if the bridged channel has a specific music class set that
455 was done in the dialplan using Set(CHANNEL(musicclass)=something).
457 The "mohinterpret" option is similar to the old "musicclass" option. It
458 specifies which music on hold class this channel would like to listen to when
459 put on hold. This music class is only effective if this channel has no music
460 class set on it from the dialplan and the bridged channel putting this one on
461 hold had no "mohsuggest" setting.
463 The IAX2 and Zap channel drivers have an additional feature for the
464 "mohinterpret" option. If this option is set to "passthrough", then these
465 channel drivers will pass through the HOLD message in signalling instead of
466 starting music on hold on the channel. An example for how this would be
467 useful is in an enterprise network of Asterisk servers. When one phone on one
468 server puts a phone on a different server on hold, the remote server will be
469 responsible for playing the hold music to its local phone that was put on
470 hold instead of the far end server across the network playing the music.
474 * The behavior of the "clid" field of the CDR has always been that it will
475 contain the callerid ANI if it is set, or the callerid number if ANI was not
476 set. When using the "callerid" option for various channel drivers, some
477 would set ANI and some would not. This has been cleared up so that all
478 channel drivers set ANI. If you would like to change the callerid number
479 on the channel from the dialplan and have that change also show up in the
480 CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
484 * There are some API functions that were not previously prefixed with the 'ast_'
485 prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
486 have a module that uses the services provided by res_adsi, res_odbc, or
487 res_agi, you will need to add ast_ prefixes to the functions that you call
492 * format_wav: The GAIN preprocessor definition has been changed from 2 to 0
493 in Asterisk 1.4. This change was made in response to user complaints of
494 choppiness or the clipping of loud signal peaks. The GAIN preprocessor
495 definition will be retained in Asterisk 1.4, but will be removed in a
496 future release. The use of GAIN for the increasing of voicemail message
497 volume should use the 'volgain' option in voicemail.conf
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