diff -Nur -x '*.orig' -x '*.rej' smeserver-freepbx-0.1/createlinks mezzanine_patched_smeserver-freepbx-0.1/createlinks
--- smeserver-freepbx-0.1/createlinks	2010-09-21 20:22:34.000000000 +0200
+++ mezzanine_patched_smeserver-freepbx-0.1/createlinks	2010-09-21 20:21:20.000000000 +0200
@@ -31,6 +31,7 @@
 
 my $event = 'freepbx-update';
 templates2events("/etc/amportal.conf", qw(freepbx-update bootstrap-console-save));
+templates2events("/etc/asterisk/asterisk.conf", qw(freepbx-update bootstrap-console-save));
 templates2events("/etc/asterisk/cdr_mysql.conf", qw(freepbx-update bootstrap-console-save));
 templates2events("/etc/asterisk/manager.conf", qw(freepbx-update bootstrap-console-save));
 templates2events("/etc/httpd/conf/httpd.conf", $event);
diff -Nur -x '*.orig' -x '*.rej' smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/10dirs mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/10dirs
--- smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/10dirs	1970-01-01 01:00:00.000000000 +0100
+++ mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/10dirs	2010-09-21 20:21:20.000000000 +0200
@@ -0,0 +1,12 @@
+[directories]
+astetcdir => /etc/asterisk
+astmoddir => { return '/usr/lib64/asterisk/modules' if -d '/usr/lib64/asterisk/modules';
+               return '/usr/lib/asterisk/modules' if -d '/usr/lib/asterisk/modules'; }
+astvarlibdir => /var/lib/asterisk
+astdatadir => /usr/share/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+
+
diff -Nur -x '*.orig' -x '*.rej' smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/20options mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/20options
--- smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/20options	1970-01-01 01:00:00.000000000 +0100
+++ mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/20options	2010-09-21 20:21:20.000000000 +0200
@@ -0,0 +1,39 @@
+[options]
+languageprefix = yes ; Use the new sound prefix path syntax
+;verbose = 3
+;debug = 3
+;alwaysfork = yes ; same as -F at startup
+;nofork = yes ; same as -f at startup
+;quiet = yes ; same as -q at startup
+;timestamp = yes ; same as -T at startup
+execincludes = yes ; support #exec in config files
+;console = yes ; Run as console (same as -c at startup)
+highpriority = yes ; Run realtime priority (same as -p at startup)
+;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
+nocolor = yes ; Disable console colors
+;dontwarn = yes ; Disable some warnings
+;dumpcore = yes ; Dump core on crash (same as -g at startup)
+;internal_timing = yes
+;systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues
+;maxcalls = 10 ; Maximum amount of calls allowed
+;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit
+;cache_record_files = yes ; Cache recorded sound files to another directory during recording
+;record_cache_dir = /tmp ; Specify cache directory (used in conjunction with cache_record_files)
+;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
+;transmit_silence = yes ; Transmit silence while a channel is in a waiting state, a recording only state, or when DTMF is
+                        ; being generated.  Note that the silence internally is generated in raw signed linear format.
+                        ; This means that it must be transcoded into the native format of the channel before it can be sent
+                        ; to the device.  It is for this reason that this is optional, as it may result in requiring a
+                        ; temporary codec translation path for a channel that may not otherwise require one.
+;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly
+;runuser = asterisk ; The user to run as
+;rungroup = asterisk ; The group to run as
+;dahdichanname = yes ; Channels created by chan_dahdi will be called 'DAHDI', otherwise 'Zap'
+
+; Changing the following lines may compromise your security.
+;[files]
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = apache
+;astctl = asterisk.ctl
+
diff -Nur -x '*.orig' -x '*.rej' smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/template-begin mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/template-begin
--- smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/template-begin	1970-01-01 01:00:00.000000000 +0100
+++ mezzanine_patched_smeserver-freepbx-0.1/root/etc/e-smith/templates/etc/asterisk/asterisk.conf/template-begin	2010-09-21 20:22:14.000000000 +0200
@@ -0,0 +1,12 @@
+;------------------------------------------------------------
+;              !!DO NOT MODIFY THIS FILE!!
+; 
+; Manual changes will be lost when this file is regenerated.
+;
+; Please read the developer's guide, which is available
+; at http://www.contribs.org/development/
+;
+; Copyright (C) 1999-2006 Mitel Networks Corporation
+;------------------------------------------------------------
+
+