1) Made pbx-asterisk executable.

2) Removed a few unused templates.
3) Removed old firewall rule-creation routine from pbx-asterisk.
4) Updated text on main page.
This commit is contained in:
Iordan Iordanov 2011-10-31 01:00:22 +00:00
parent 96f494f4ba
commit 62118b369b
4 changed files with 11 additions and 59 deletions

View file

@ -75,16 +75,16 @@ end
m = Map (modulename, translate("PBX Main Page"), m = Map (modulename, translate("PBX Main Page"),
translate("This configuration page allows you to configure a phone system (PBX) service which\ translate("This configuration page allows you to configure a phone system (PBX) service which\
permits making phone calls with, and sharing multiple Google and SIP (like Sipgate,\ permits making phone calls through multiple Google and SIP (like Sipgate,\
SipSorcery, and Betamax) accounts among many SIP devices. Note that Google\ SipSorcery, and Betamax) accounts and sharing them among many SIP devices. \
accounts, SIP accounts, and local user accounts are configured in the\ Note that Google accounts, SIP accounts, and local user accounts are configured in the \
\"Google Accounts\", \"SIP Accounts\", and \"User Accounts\" sub-sections.\ \"Google Accounts\", \"SIP Accounts\", and \"User Accounts\" sub-sections. \
You must configure at least one local SIP account\ You must add at least one User Account to this PBX, and then configure a SIP device or softphone \
on this PBX, to make and receive calls with your Google/SIP accounts.\ to use the account, in order to make and receive calls with your Google/SIP accounts. \
Configuring multiple users will allow you to make free calls between users, and share the configured\ Configuring multiple users will allow you to make free calls between all users, and share the configured \
Google and SIP accounts. If you have more than one Google and SIP accounts set up,\ Google and SIP accounts. If you have more than one Google and SIP accounts set up, \
you should probably configure how calls to and from them are routed in the \"Call Routing\" page.\ you should probably configure how calls to and from them are routed in the \"Call Routing\" page. \
If you're interested in using your own PBX from anywhere in the world,\ If you're interested in using your own PBX from anywhere in the world, \
then visit the \"Remote Usage\" section in the \"Advanced Settings\" page.")) then visit the \"Remote Usage\" section in the \"Advanced Settings\" page."))
---------------------------------------------------------------------------------------------------- ----------------------------------------------------------------------------------------------------
@ -105,7 +105,7 @@ function sts.cfgvalue(self, section)
usrs = luci.sys.exec("asterisk -rx 'sip show users'") usrs = luci.sys.exec("asterisk -rx 'sip show users'")
chan = luci.sys.exec("asterisk -rx 'core show channels'") chan = luci.sys.exec("asterisk -rx 'core show channels'")
return format_two_indices(reg, 1, 5) .. format_two_indices(jab, 2, 4) .. "\n" return format_two_indices(reg, 1, 5) .. format_two_indices(jab, 2, 4) .. "\n"
.. format_one_index(usrs,1) .. "\n" .. chan .. format_one_index(usrs, 1) .. "\n" .. chan
elseif server == "freeswitch" then elseif server == "freeswitch" then
return "Freeswitch is not supported yet.\n" return "Freeswitch is not supported yet.\n"
else else

44
applications/luci-pbx/root/etc/init.d/pbx-asterisk Normal file → Executable file
View file

@ -33,8 +33,6 @@ ASTGROUP=nogroup
ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk" ASTDIRSRECURSIVE="/var/run/asterisk /var/log/asterisk /var/spool/asterisk"
ASTDIRS="/usr/lib/asterisk" ASTDIRS="/usr/lib/asterisk"
FIREWALL_PATH="/etc/asterisk/firewall.$MODULENAME"
TEMPLATEDIR=/etc/${MODULENAME}-asterisk TEMPLATEDIR=/etc/${MODULENAME}-asterisk
ASTERISKDIR=/etc/asterisk ASTERISKDIR=/etc/asterisk
WORKDIR=/tmp/$MODULENAME.$$ WORKDIR=/tmp/$MODULENAME.$$
@ -73,7 +71,6 @@ TMPL_EXTOUTGTALK=$TEMPLATEDIR/extensions_outgoing_gtalk.conf.TEMPLATE
TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE TMPL_EXTOUTLOCAL=$TEMPLATEDIR/extensions_outgoing_dial_local_user.conf.TEMPLATE
TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE TMPL_EXTOUTSIP=$TEMPLATEDIR/extensions_outgoing_sip.conf.TEMPLATE
TMPL_FIREWALL=$TEMPLATEDIR/firewall.$MODULENAME.TEMPLATE
TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE TMPL_JABBER=$TEMPLATEDIR/jabber.conf.TEMPLATE
TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE TMPL_JABBERUSER=$TEMPLATEDIR/jabber_users.conf.TEMPLATE
TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE TMPL_SIP=$TEMPLATEDIR/sip.conf.TEMPLATE
@ -557,47 +554,6 @@ pbx_fix_ownership()
chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE
} }
# Creates firewall configuration. However, since this functionality is now
# taken over by pbx-advanced.lua, this function is unused.
create_firewall_config()
{
local bindport
local rtpstart
local rtpend
local externhost
config_get bindport advanced bindport
config_get externhost advanced externhost
config_get rtpstart advanced rtpstart
config_get rtpend advanced rtpend
# We need all of these parameters to be set in order to consider inserting firewall rules.
if [ -z "$externhost" -o -z "$bindport" -o -z "$rtpstart" -o -z "$rtpend" ] ; then
rm -f $FIREWALL_PATH
return
fi
sed "s/|SIPPORT|/$bindport/g" $TMPL_FIREWALL |\
sed "s/|RTPRANGE|/$rtpstart:$rtpend/g" > $WORKDIR/firewall.$MODULENAME
# Make sure there is an include section in the firewall configuration
# to include the file we just created.
i=0 ; found_path=0
while p=`uci get firewall.@include[$i].path 2>/dev/null` ; do
if [ "$p" = "$FIREWALL_PATH" ] ; then
found_path=1
break
fi
i=`expr $i + 1`
done
# If no include section was found which mentions $FIREWALL_PATH, add one.
if [ $found_path -eq 0 ] ; then
uci add firewall include 1>/dev/null 2>/dev/null
uci set firewall.@include[-1].path="$FIREWALL_PATH"
uci commit firewall
fi
}
start() { start() {
mkdir -p $WORKDIR mkdir -p $WORKDIR

View file

@ -1,2 +0,0 @@
iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |SIPPORT| -j ACCEPT
iptables -I INPUT 3 -p udp -s 0/0 -d 0/0 --dport |RTPRANGE| -j ACCEPT