Changeset 7529
- Timestamp:
- 03/14/09 07:11:39 (4 years ago)
- Files:
-
- freepbx/trunk/amp_conf/astetc/logger.conf (modified) (1 diff)
- freepbx/trunk/amp_conf/astetc/manager.conf (modified) (1 diff)
- freepbx/trunk/amp_conf/astetc/modules.conf (modified) (2 diffs)
- freepbx/trunk/amp_conf/astetc/rtp.conf (modified) (1 diff)
Legend:
- Unmodified
- Added
- Removed
- Modified
- Copied
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freepbx/trunk/amp_conf/astetc/logger.conf
r8 r7529 26 26 ; 27 27 ;debug => debug 28 ; The DTMF log is very handy if you have issues with IVR's 29 ;dtmf => dtmf 28 30 ;console => notice,warning,error 29 31 ;console => notice,warning,error,debug freepbx/trunk/amp_conf/astetc/manager.conf
r1785 r7529 1 1 ; 2 ; Asterisk Call Management support 2 ; AMI - Asterisk Manager interface 3 ; 4 ; FreePBX needs this to be enabled. Note that if you enable it on a different IP, you need 5 ; to assure that this can't be reached from un-authorized hosts with the ACL settings (permit/deny). 6 ; Also, remember to configure non-default port or IP-addresses in amportal.conf. 7 ; 8 ; The AMI connection is used both by the portal and the operator's panel in FreePBX. 9 ; 10 ; FreePBX assumes an AMI connection to localhost:5038 by default. 3 11 ; 4 12 [general] freepbx/trunk/amp_conf/astetc/modules.conf
r7345 r7529 1 1 ; 2 ; Asterisk configuration file2 ; Asterisk Module Loader configuration file 3 3 ; 4 ; Module Loader configuration file5 4 ; 6 5 … … 36 35 load => format_pcm.so 37 36 load => format_au.so 38 ; This isn't part of ' asterisk' iteslf, it's part of asterisk-addons. If this isn't37 ; This isn't part of 'Asterisk' iteslf, it's part of asterisk-addons. If this isn't 39 38 ; installed, asterisk will fail to start. But it does need to go here for native MOH 40 39 ; to work using mp3's. 40 ; Note that on a system with a high number of calls, using a compressed audio format for 41 ; musiconhold takes CPU resources. Converting these files to ulaw/alaw makes the job 42 ; much easier for your CPU. 41 43 load => format_mp3.so 42 44 load => res_musiconhold.so 43 45 ; 44 46 ; Load either OSS or ALSA, not both 45 ; By default, load OSS only (automatically) and do not load ALSA47 ; By default, load no console driver 46 48 ; 47 49 noload => chan_alsa.so 48 50 noload => chan_oss.so 49 51 ; 50 ; Module names listed in "global" section will have symbols globally51 ; exported to modules loaded after them.52 ;53 52 noload => app_directory_odbcstorage.so 54 53 noload => app_voicemail_odbcstorage.so 55 56 [global]freepbx/trunk/amp_conf/astetc/rtp.conf
r1725 r7529 5 5 ; 6 6 ; RTP start and RTP end configure start and end addresses 7 ; These are the addresses where your system will RECEIVE audio and video streams. 8 ; If you have connections across a firewall, make sure that these are open. 7 9 ; 8 10 rtpstart=10000
