aboutsummaryrefslogtreecommitdiffstats
path: root/UPGRADE.txt
blob: e804548f210b3cc54a764cc87af08daf1bf54f3d (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
===========================================================
===
=== Information for upgrading between Asterisk 1.6 versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
===
===========================================================

From 1.6.1.1 to 1.6.1.2:

* Beginning with this release, Asterisk's internal methods of
  negotiating T.38 (FAX over IP) sessions changed in
  non-backwards-compatible ways. Any applications that previously used
  AST_CONTROL_T38 control frames will have to be upgraded to use
  AST_CONTROL_T38_PARAMETERS control frames instead; app_fax.c is a good
  example of how to generate and respond to these frames. These changes
  were made to solve significant T.38 interoperability problems between
  Asterisk and various SIP/T.38 endpoints identified by many users of
  Asterisk.

From 1.6.0.1 to 1.6.1:

* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
  API calls were added in 1.6.0, so that modules that provide multiple
  AGI commands could register/unregister them all with a single
  step. However, these API calls were not implemented properly, and did
  not allow the caller to know whether registration or unregistration
  succeeded or failed. They have been redefined to now return success
  or failure, but this means any code using these functions will need
  be recompiled after upgrading to a version of Asterisk containing
  these changes. In addition, the source code using these functions
  should be reviewed to ensure it can properly react to failure
  of registration or unregistration of its API commands.

* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
  to better match what it really does, and the argument order has been
  changed to be consistent with other API calls that perform similar
  operations.

From 1.6.0.x to 1.6.1:

* In previous versions of Asterisk, due to the way objects were arranged in
  memory by chan_sip, the order of entries in sip.conf could be adjusted to
  control the behavior of matching against peers and users.  The way objects
  are managed has been significantly changed for reasons involving performance
  and stability.  A side effect of these changes is that the order of entries
  in sip.conf can no longer be relied upon to control behavior.

* The following core commands dealing with dialplan have been deprecated: 'core
  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
  instead.

* In the dialplan expression parser, the logical value of spaces
  immediately preceding a standalone 0 previously evaluated to
  true. It now evaluates to false.  This has confused a good many
  people in the past (typically because they failed to realize the
  space had any significance).  Since this violates the Principle of
  Least Surprise, it has been changed.

* While app_directory has always relied on having a voicemail.conf or users.conf file
  correctly set up, it now is dependent on app_voicemail being compiled as well.

* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
  and you should start using that function instead for retrieving information about
  the channel in a technology-agnostic way.

* If you have any third party modules which use a config file variable whose
  name ends in a '+', please note that the append capability added to this
  version may now conflict with that variable naming scheme.  An easy
  workaround is to ensure that a space occurs between the '+' and the '=',
  to differentiate your variable from the append operator.  This potential
  conflict is unlikely, but is documented here to be thorough.

* The "Join" event from app_queue now uses the CallerIDNum header instead of
  the CallerID header to indicate the CallerID number.

* Support for Taiwanese was incorrectly supported with the "tw" language code.
  In reality, the "tw" language code is reserved for the Twi language, native
  to Ghana.  If you were previously using the "tw" language code, you should
  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
  specific localizations.  Additionally, "mx" should be changed to "es_MX",
  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
  "cs", not "cz".

* If you use ODBC storage for voicemail, there is a new field called "flag"
  which should be a char(8) or larger.  This field specifies whether or not a
  message has been designated to be "Urgent", "PRIORITY", or not.

From 1.6.1.1 to 1.6.1.2:

* Support for Taiwanese was incorrectly supported with the "tw" language code.
  In reality, the "tw" language code is reserved for the Twi language, native
  to Ghana.  If you were previously using the "tw" language code, you should
  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
  specific localizations.  Additionally, "mx" should be changed to "es_MX",
  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
  "cs", not "cz".