root/freepbx/branches/2.10/amp_conf/astetc/modules.conf

Revision 9043, 2.3 kB (checked in by p_lindheimer, 3 years ago)

Merged revisions 8714-9042 via svnmerge from
http://svn.freepbx.org/freepbx/branches/2.7

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r8714 | p_lindheimer | 2010-01-30 08:02:33 -0800 (Sat, 30 Jan 2010) | 1 line


branch trunk to 2.7

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r8731 | p_lindheimer | 2010-01-30 09:18:30 -0800 (Sat, 30 Jan 2010) | 1 line


Creating release 2.7.0beta1

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r8750 | mickecarlsson | 2010-02-01 13:25:40 -0800 (Mon, 01 Feb 2010) | 1 line


Closes #2839 #3980 #3992 added setting in amportal.conf for using Google DNS for enumlookup.ago

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r8751 | mickecarlsson | 2010-02-01 13:29:00 -0800 (Mon, 01 Feb 2010) | 1 line


Dont use language specific page in the url

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r8763 | mickecarlsson | 2010-02-02 11:02:41 -0800 (Tue, 02 Feb 2010) | 1 line


Updated amp.pot for 2.7

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r8764 | mickecarlsson | 2010-02-02 11:25:42 -0800 (Tue, 02 Feb 2010) | 1 line


Spelling error fix

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r8766 | mickecarlsson | 2010-02-02 11:46:04 -0800 (Tue, 02 Feb 2010) | 1 line


Spelling error fixes for 2.7 branch

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r8772 | mickecarlsson | 2010-02-02 14:34:58 -0800 (Tue, 02 Feb 2010) | 1 line


Close #4024 add check for Asterisk 1.6 in GotoIfTime? extensions class

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r8829 | p_lindheimer | 2010-02-16 08:09:12 -0800 (Tue, 16 Feb 2010) | 9 lines


Merged revisions 8828 via svnmerge from
http://svn.freepbx.org/freepbx/branches/2.6


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r8828 | p_lindheimer | 2010-02-16 08:06:59 -0800 (Tue, 16 Feb 2010) | 1 line


sytax error == should be assignment =

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r8843 | p_lindheimer | 2010-02-17 09:17:43 -0800 (Wed, 17 Feb 2010) | 1 line


change mohmp3 to moh, put moh as default in template, use mohmp3 as fallback default when not specified for compatibility re #4051

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r8846 | p_lindheimer | 2010-02-17 12:15:05 -0800 (Wed, 17 Feb 2010) | 1 line


closes #4052 add option to force reinstall or downgrade modules

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r8847 | p_lindheimer | 2010-02-17 12:23:23 -0800 (Wed, 17 Feb 2010) | 1 line


bump the base version to RC1 in prep for RC1 tarball

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r8848 | p_lindheimer | 2010-02-17 12:27:26 -0800 (Wed, 17 Feb 2010) | 1 line


update CHANGES for RC release

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r8849 | p_lindheimer | 2010-02-17 12:36:58 -0800 (Wed, 17 Feb 2010) | 1 line


closes #3575 check for ASTMANAGERHOST and ASTMANAGERPORT

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r8860 | p_lindheimer | 2010-02-17 16:50:54 -0800 (Wed, 17 Feb 2010) | 1 line


Creating release 2.7.0RC1

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r8861 | p_lindheimer | 2010-02-17 16:53:25 -0800 (Wed, 17 Feb 2010) | 1 line


Modify build tools to deal with change from mohmp3 to moh

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r8862 | p_lindheimer | 2010-02-17 16:53:45 -0800 (Wed, 17 Feb 2010) | 1 line


Creating release 2.7.0RC1

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r8873 | p_lindheimer | 2010-02-17 17:10:45 -0800 (Wed, 17 Feb 2010) | 1 line


remove the wihtmodules generation part of script, we don't use that anymore

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r8880 | mickecarlsson | 2010-02-18 13:54:51 -0800 (Thu, 18 Feb 2010) | 1 line


Finally, an automated tool to generate all .pot files for localization, ugly but working. Requires some setup yet to be documented, should have merge from trunk but I did not manage that so here is another ci

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r8881 | p_lindheimer | 2010-02-18 16:45:33 -0800 (Thu, 18 Feb 2010) | 1 line


fixes #4057 don't iterate through global array when calling hooks or hooks that do the same will end the loop prematurely

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r8907 | p_lindheimer | 2010-02-22 16:59:53 -0800 (Mon, 22 Feb 2010) | 1 line


adds un-published option, USEDIALONE, to use the experimental macro-dial-one in place of macro-dial + dialparties.agi when dialing single extensions. The macro has been very little tested and is otherwise not used at all if this is not set. However, this will enable for those who want to start to test the macro and help flush out its operation for a 2.8 target re #4068

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r8923 | mickecarlsson | 2010-02-24 12:38:40 -0800 (Wed, 24 Feb 2010) | 1 line


Commented out format_au.so as it is removed from Asterisk 1.4 and later.

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r8950 | mickecarlsson | 2010-02-27 06:40:10 -0800 (Sat, 27 Feb 2010) | 1 line


Updated amp.pot, updated swedish language

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r8952 | mickecarlsson | 2010-02-27 07:45:55 -0800 (Sat, 27 Feb 2010) | 1 line


Removed obsolete language tool

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r8971 | p_lindheimer | 2010-02-27 21:48:35 -0800 (Sat, 27 Feb 2010) | 1 line


2.7.0 upgrade directory for release

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r8972 | p_lindheimer | 2010-02-27 21:52:36 -0800 (Sat, 27 Feb 2010) | 1 line


update CHANGES

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r9029 | p_lindheimer | 2010-02-27 22:17:38 -0800 (Sat, 27 Feb 2010) | 1 line


Creating release 2.7.0

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  • Property svn:eol-style set to native
  • Property svn:keywords set to Author Date Id Revision
Line 
1 ;
2 ; Asterisk Module Loader configuration file
3 ;
4 ;
5
6 [modules]
7 autoload=yes
8 ;
9 ; Any modules that need to be loaded before the Asterisk core has been
10 ; initialized (just after the logger has been initialized) can be loaded
11 ; using 'preload'. This will frequently be needed if you wish to map all
12 ; module configuration files into Realtime storage, since the Realtime
13 ; driver will need to be loaded before the modules using those configuration
14 ; files are initialized.
15 ;
16 ; An example of loading ODBC support would be:
17 ;preload => res_odbc.so
18 ;preload => res_config_odbc.so
19 ;
20 ; As FreePBX is using Local as the channel for queue members we need to make sure
21 ; that pbx_config.so and chan_local.so are preloaded. If not, queue members
22 ; will be marked as invalid until app_queue is reloaded.
23 preload => pbx_config.so
24 preload => chan_local.so
25 ;
26 ; Uncomment the following if you wish to use the Speech Recognition API
27 ;preload => res_speech.so
28 ;
29 ; If you want, load the GTK console right away. 
30 ; KDE console is obsolete and was removed from Asterisk 2008-01-10
31 ;
32 noload => pbx_gtkconsole.so
33 ;load => pbx_gtkconsole.so
34 noload => pbx_kdeconsole.so
35 ;
36 ; Intercom application is obsoleted by
37 ; chan_oss.  Don't load it.
38 ;
39 noload => app_intercom.so
40 ;
41 ; DON'T load the chan_modem.so, as they are obsolete in * 1.2
42
43 noload => chan_modem.so
44 noload => chan_modem_aopen.so
45 noload => chan_modem_bestdata.so
46 noload => chan_modem_i4l.so
47
48 ; Trunkisavail is a broken module supplied by Trixbox
49 noload => app_trunkisavail.so
50
51 ; Ensure that format_* modules are loaded before res_musiconhold
52 ;load => format_ogg_vorbis.so
53 load => format_wav.so
54 load => format_pcm.so
55
56 ; format_au.so is removed from Asterisk 1.4 and later, remove ; to enable
57 ;load => format_au.so
58
59 ; This isn't part of 'Asterisk' iteslf, it's part of asterisk-addons. If this isn't
60 ; installed, asterisk will fail to start. But it does need to go here for native MOH
61 ; to work using mp3's.
62 ;   Note that on a system with a high number of calls, using a compressed audio format for
63 ;   musiconhold takes CPU resources. Converting these files to ulaw/alaw makes the job
64 ;   much easier for your CPU.
65 load => format_mp3.so
66 load => res_musiconhold.so
67 ;
68 ; Load either OSS or ALSA, not both
69 ; By default, load no console driver
70 ;
71 noload => chan_alsa.so
72 noload => chan_oss.so
73 ;
74 noload => app_directory_odbcstorage.so
75 noload => app_voicemail_odbcstorage.so
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