[Astpp-commit] SF.net SVN: astpp:[2279] trunk/samples

Revision: 2279
          http://astpp.svn.sourceforge.net/astpp/?rev=2279&view=rev
Author:   darrenkw
Date:     2009-10-06 03:42:26 +0000 (Tue, 06 Oct 2009)

Log Message:
-----------
Removed old sample files.

Removed Paths:
-------------
    trunk/samples/sample.astpp-enh-config.conf
    trunk/samples/sample.reseller-config.conf

Deleted: trunk/samples/sample.reseller-config.conf
===================================================================
--- trunk/samples/sample.reseller-config.conf	2009-10-06 03:41:14 UTC (rev 2278)
+++ trunk/samples/sample.reseller-config.conf	2009-10-06 03:42:26 UTC (rev 2279)
@@ -1,118 +0,0 @@
-# This file contains the more advanced ASTPP variables which should remain
-# mostly constant between installs and which you do not want to change too easily.
-# Changing this variables without knowing exactly what you are doing could have
-# far ranging consequences.
-# The Author
-
-results_per_page = 30 # How many results per page do we should in the web interface?
-astpp_dir = /var/lib/astpp # Where do the astpp configs live?
-
-# This is the override authorization code and will allow access to the system.  Should
-# be set to something "impossible" to guess. DO NOT LEAVE AT THE DEFAULT!!!
-auth = a23asudf9810-zalkj32423
-
-# Database type:  ASTPP was designed for a MySQL database initially.  Valid options are:
-# MySQL. Pgsql is coming but is not ready yet.
-astpp_dbengine = MySQL
-rt_dbengine = MySQL
-cdr_dbengine = MySQL
-osc_dbengine = MySQL
-agile_dbengine = MySQL
-freepbx_dbengine = MySQL
-
-# Please specify the external billing application to use.  If you are not using any
-# the leave it blank.  Valid options are "agile" and "oscommerce".
-externalbill = oscommerce
-
-# Do you wish to enable calling cards?  1 for yes and 2 for no.
-callingcards = 1
-
-# Change this one at your own peril.  If you switch it off, calls will not be marked
-# as billed in asterisk once they are billed.
-astcdr = 1
-
-# Change this one at your own peril.  If you switch it off, calls will not be written
-# to astpp when they are calculated.
-posttoastpp = 1
-
-# This is used when calling astpp-rate-engine.pl from the extensions.conf file.
-# I would recommend 10 seconds as that gives that time to Asterisk to get the call written
-# to the cdr database.
-sleep = 10
-
-# If this is enabled, ASTPP will create users and DIDs in the FreePBX (www.freepbx.org)
-# mysql DB.  
-users_dids_amp = 0
-
-# If this is enabled, ASTPP will create users and DIDs in the Asterisk Realtime
-# mysql DB. 
-users_dids_rt = 1
-
-# Service prepend is the number that ASTPP attaches to the front of the id that it is passed
-# in astpp-auto-admin.cgi
-# If service_prepend is left blank, then the new method kicks in.  The new method allows you
-# to specify a required extension length and default filler.  It then chops the strings up.
-service_prepend = 778
-service_length = 7
-service_filler = 4110000
-
-# AgileBill(Trademark of AgileCo) Settings:
-agile_host = 127.0.0.1
-agile_db = agile
-agile_user = root
-agile_pass = 
-agile_site_id = 1
-agile_charge_status = 0
-agile_taxable = 1
-agile_dbprefix = _
-agile_service_prepend = 778
-
-# OSCommerce Settings (www.oscommerce.org)
-osc_host = 127.0.0.1
-osc_db = oscommerce
-osc_user = root
-osc_pass = password
-osc_product_id = 99999999
-osc_payment_method = "Charge"
-osc_order_status = 1
-
-# FreePBX Settings (www.freepbx.org)
-freepbx_host = 127.0.0.1
-freepbx_db = asterisk
-freepbx_user = root
-freepbx_pass = passw0rd
-freepbx_iax_table = iax
-freepbx_sip_table = sip
-freepbx_extensions_table = extensions
-freepbx_codec_allow = g729,ulaw,alaw
-freepbx_codec_disallow = all
-freepbx_mailbox_group = default
-freepbx_sip_nat = yes
-freepbx_sip_canreinvite = no
-freepbx_sip_dtmfmode = rfc2833
-freepbx_sip_qualify = yes
-freepbx_sip_type = friend
-freepbx_sip_callgroup = 
-freepbx_sip_pickupgroup = 
-freepbx_iax_notransfer = yes
-freepbx_iax_type = friend
-freepbx_iax_qualify = yes
-
-# Asterisk -realtime Settings
-rt_host = 127.0.0.1
-rt_db = realtime
-rt_user = root
-rt_pass = 
-rt_iax_table = iax
-rt_sip_table = sip
-rt_extensions_table = extensions
-rt_sip_insecure = very
-rt_sip_nat = yes
-rt_sip_canreinvite = no
-rt_codec_allow = g729,ulaw,alaw
-rt_codec_disallow = all
-rt_mailbox_group = default
-rt_sip_qualify = yes
-rt_sip_type = friend
-rt_iax_qualify = yes
-rt_voicemail_table = voicemail_users


This was sent by the SourceForge.net collaborative development platform, the world's largest Open Source development site.