1 ===========================================================
3 === Information for upgrading between Asterisk versions
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
16 === UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
17 === UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
19 ===========================================================
23 * Asterisk-addons no longer exists as an independent package. Those modules
24 now live in the addons directory of the main Asterisk source tree. They
25 are not enabled by default. For more information about why modules live in
26 addons, see README-addons.txt.
28 * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
29 users of this channel in the tree have been converted to LOG_NOTICE or removed
30 (in cases where the same message was already generated to another channel).
32 * The usage of RTP inside of Asterisk has now become modularized. This means
33 the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
34 If you are not using autoload=yes in modules.conf you will need to ensure
35 it is set to load. If not, then any module which uses RTP (such as chan_sip)
36 will not be able to send or receive calls.
38 * The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
39 remains. It now exists within app_chanspy.c and retains the exact same
40 functionality as before.
42 * The default behavior for Set, AGI, and pbx_realtime has been changed to implement
43 1.6 behavior by default, if there is no [compat] section in asterisk.conf. In
44 prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
45 Specifically, that means that pbx_realtime and res_agi expect you to use commas
46 to separate arguments in applications, and Set only takes a single pair of
47 a variable name/value. The old 1.4 behavior may still be obtained by setting
48 app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
51 * The PRI channels in chan_dahdi can no longer change the channel name if a
52 different B channel is selected during call negotiation. To prevent using
53 the channel name to infer what B channel a call is using and to avoid name
54 collisions, the channel name format is changed.
55 The new channel naming for PRI channels is:
56 DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
58 * The ChanIsAvail application has been changed so the AVAILSTATUS variable
59 no longer contains both the device state and cause code. The cause code
60 is now available in the AVAILCAUSECODE variable. If existing dialplan logic
61 is written to expect AVAILSTATUS to contain the cause code it needs to be
62 changed to use AVAILCAUSECODE.
64 * ExternalIVR will now send Z events for invalid or missing files, T events
65 now include the interrupted file and bugs in argument parsing have been
66 fixed so there may be arguments specified in incorrect ways that were
67 working that will no longer work.
68 Please see doc/externalivr.txt for details.
70 * OSP lookup application changes following variable names:
71 OSPPEERIP to OSPINPEERIP
73 OSPDEST to OSPDESTINATION
74 OSPCALLING to OSPOUTCALLING
75 OSPCALLED to OSPOUTCALLED
76 OSPRESULTS to OSPDESTREMAILS
78 * The Manager event 'iax2 show peers' output has been updated. It now has a
79 similar output of 'sip show peers'.
81 * VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
82 of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
83 the current dialplan context.
85 * Environment variables that start with "AST_" are reserved to the system and
86 may no longer be set from the dialplan.
88 * When a call is redirected inside of a Dial, the app and appdata fields of the
89 CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
91 * The CDR handling of billsec and duration field has changed. If your table
92 definition specifies those fields as float,double or similar they will now
93 be logged with microsecond accuracy instead of a whole integer.
95 * chan_sip will no longer set up a local call forward when receiving a
96 482 Loop Detected response. The dialplan will just continue from where it
101 * SIP no longer sends the 183 progress message for early media by
102 default. Applications requiring early media should use the
103 progress() dialplan app to generate the progress message.
105 * The firmware for the IAXy has been removed from Asterisk. It can be
106 downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk
107 install the firmware into its proper location, place the firmware in the
108 contrib/firmware/iax/ directory in the Asterisk source tree before running
111 * T.38 FAX error correction mode can no longer be configured in udptl.conf;
112 instead, it is configured on a per-peer (or global) basis in sip.conf, with
113 the same default as was present in udptl.conf.sample.
115 * T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
116 instead, it is either supplied by the application servicing the T.38 channel
117 (for a FAX send or receive) or calculated from the bridged endpoint's
118 maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
119 allows for overriding the value supplied by a remote endpoint, which is useful
120 when T.38 connections are made to gateways that supply incorrectly-calculated
121 maximum datagram sizes.
123 * There have been some changes to the IAX2 protocol to address the security
124 concerns documented in the security advisory AST-2009-006. Please see the
125 IAX2 security document, doc/IAX2-security.pdf, for information regarding
126 backwards compatibility with versions of Asterisk that do not contain these
129 * The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
130 has been renamed to 'directmedia', to better reflect what it actually does.
131 In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
132 starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
133 option never had any effect on these cases, it only affected the re-INVITEs
134 used for direct media path setup. For MGCP and Skinny, the option was poorly
135 named because those protocols don't even use INVITE messages at all. For
136 backwards compatibility, the old option is still supported in both normal
137 and Realtime configuration files, but all of the sample configuration files,
138 Realtime/LDAP schemas, and other documentation refer to it using the new name.
140 * The default console now will use colors according to the default background
141 color, instead of forcing the background color to black. If you are using a
142 light colored background for your console, you may wish to use the option
143 flag '-W' to present better color choices for the various messages. However,
144 if you'd prefer the old method of forcing colors to white text on a black
145 background, the compatibility option -B is provided for this purpose.
147 * SendImage() no longer hangs up the channel on transmission error or on
148 any other error; in those cases, a FAILURE status is stored in
149 SENDIMAGESTATUS and dialplan execution continues. The possible
150 return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
151 UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
152 has been replaced with 'UNSUPPORTED'). This change makes the
153 SendImage application more consistent with other applications.
155 * skinny.conf now has separate sections for lines and devices.
156 Please have a look at configs/skinny.conf.sample and update
159 * Queue names previously were treated in a case-sensitive manner,
160 meaning that queues with names like "sales" and "sALeS" would be
161 seen as unique queues. The parsing logic has changed to use
162 case-insensitive comparisons now when originally hashing based on
163 queue names, meaning that now the two queues mentioned as examples
164 earlier will be seen as having the same name.
166 * The SPRINTF() dialplan function has been moved into its own module,
167 func_sprintf, and is no longer included in func_strings. If you use this
168 function and do not use 'autoload=yes' in modules.conf, you will need
169 to explicitly load func_sprintf for it to be available.
171 * The res_indications module has been removed. Its functionality was important
172 enough that most of it has been moved into the Asterisk core.
173 Two applications previously provided by res_indications, PlayTones and
174 StopPlayTones, have been moved into a new module, app_playtones.
176 * Support for Taiwanese was incorrectly supported with the "tw" language code.
177 In reality, the "tw" language code is reserved for the Twi language, native
178 to Ghana. If you were previously using the "tw" language code, you should
179 switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
180 specific localizations. Additionally, "mx" should be changed to "es_MX",
181 Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
184 * DAHDISendCallreroutingFacility() parameters are now comma-separated,
185 instead of the old pipe.
187 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration
188 that would end up being interpreted as a bug once Asterisk started removing
189 the contacts from a user list.
191 * The cdr.conf file must exist and be configured correctly in order for CDR
192 records to be written.
194 From 1.6.0.1 to 1.6.1:
196 * The ast_agi_register_multiple() and ast_agi_unregister_multiple()
197 API calls were added in 1.6.0, so that modules that provide multiple
198 AGI commands could register/unregister them all with a single
199 step. However, these API calls were not implemented properly, and did
200 not allow the caller to know whether registration or unregistration
201 succeeded or failed. They have been redefined to now return success
202 or failure, but this means any code using these functions will need
203 be recompiled after upgrading to a version of Asterisk containing
204 these changes. In addition, the source code using these functions
205 should be reviewed to ensure it can properly react to failure
206 of registration or unregistration of its API commands.
208 * The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
209 to better match what it really does, and the argument order has been
210 changed to be consistent with other API calls that perform similar
213 From 1.6.0.x to 1.6.1:
215 * In previous versions of Asterisk, due to the way objects were arranged in
216 memory by chan_sip, the order of entries in sip.conf could be adjusted to
217 control the behavior of matching against peers and users. The way objects
218 are managed has been significantly changed for reasons involving performance
219 and stability. A side effect of these changes is that the order of entries
220 in sip.conf can no longer be relied upon to control behavior.
222 * The following core commands dealing with dialplan have been deprecated: 'core
223 show globals', 'core set global' and 'core set chanvar'. Use the equivalent
224 'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
227 * In the dialplan expression parser, the logical value of spaces
228 immediately preceding a standalone 0 previously evaluated to
229 true. It now evaluates to false. This has confused a good many
230 people in the past (typically because they failed to realize the
231 space had any significance). Since this violates the Principle of
232 Least Surprise, it has been changed.
234 * While app_directory has always relied on having a voicemail.conf or users.conf file
235 correctly set up, it now is dependent on app_voicemail being compiled as well.
237 * SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
238 and you should start using that function instead for retrieving information about
239 the channel in a technology-agnostic way.
241 * If you have any third party modules which use a config file variable whose
242 name ends in a '+', please note that the append capability added to this
243 version may now conflict with that variable naming scheme. An easy
244 workaround is to ensure that a space occurs between the '+' and the '=',
245 to differentiate your variable from the append operator. This potential
246 conflict is unlikely, but is documented here to be thorough.
248 * The "Join" event from app_queue now uses the CallerIDNum header instead of
249 the CallerID header to indicate the CallerID number.
251 * If you use ODBC storage for voicemail, there is a new field called "flag"
252 which should be a char(8) or larger. This field specifies whether or not a
253 message has been designated to be "Urgent", "PRIORITY", or not.