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
|
===========================================================
===
=== 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 1.6.0.1 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.
|