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Billion 800VGT Router
<@
PSTN>:
This
refers
to
making
calls
via
the
PSTN
line.
<@ENUM>:
This
referes
to
making
a
VoIP
SIP
call,
via
an
E.164
number
(“ENUM”),
to
an
ENUM
callee.
Electronic
Number
(ENUM)
is
a
system
that
uses
DNS
(Domain
Network
System)
based
technology
to
map
between
a
traditional
phone
number
(PSTN)
and
an
Internet
address/
SIP
URL.
The
ENUM
number
must
be
registered
on
a
public
ENUM
site
or
with
your
VoIP
Service
Provider.
<@
SIPgateway>:
This
is
used
when
using
the
Intelligent
Call
Routing
feature.
You
will
need
to
set
up
your
SIP
account
on
the
VoIP
User-defined
Profiles
link
on
the
VoIP
Wizard
page.
Go
to
the
VoIP
Wizard
in
this
manual
for
more
information.
Dial-Plan
Examples:
Description
x
.
Any
sequence of digits between
0
and
9.
Maximum
sequence
length
is
16.
xxx
Any
3
digit
(between
0
and
9)
number.
Total
sequence
length
is
3.
Note:
No
period
is
needed
(.)
xxxx
.
Any
sequence
of
digits
(between
0
and
9
)
between
3
an
16
digits
long
.
123x
.
Any
number
(0-9)
starting
with
123.
Maximum
length is
16.
[x…x]x.
For
example:
[124]x
.
Any
number
(0-9)
starting
with
1
or
2
or
4.
Maximum
length
is
16.
[x-x]x
.
For
example:
[1-3]x
.
Any
number
(0-9)
starting
with
a
number
between
1
and
3.
Maximum
length
is
16.
x[x-x]x.
For
example:
9[4-6]8x
.
Any
number
(0-9)
starting
with
9,
the
second
number
being
any
number
between
4-6,
and
the
third
number
being
8. Maximum
length
is
16.
Special
Dial
Plan
Examples:
Description
*
xx
*
x
.
Starting
with
an
asterisk,
then
any
two
digit
number,
a
‘*
sign’,
and
any
number
of
additional
digits.
Maximum
length
is
16.
*
xx
Starting
with
an
asterisk,
then
any
two
digit
number
between
00
and
99.
Total
length
including
the
*
is
3
digits.
Note:
No
period
is
needed
(.)
**
xx
*
x
.
Starting
with
two
asterisks,
then
any
two
digit
number,
a
‘*
sign’
and
any
number
of
additional
digits.
Maximum
length
is
16..
#
xx
.
Starting
with
‘#
sign’,
followed
by
any
number
with
a
length
of
between
1
and
16
digits.
##
xx
*
x
.
Starting
with
‘##
sign’
,
and
followed
by
any
two
digit
number,
a ‘*
sign’
and
ending
in
any
number
with
a
maximum
length
of
16.
<@SIPgateway>
/
Intelligent
Call
Routing
Example:
The
VoIP
Gateway
let
you
use
3
VoIP/SIP
providers
at
the
same
time.
Each
provider
has
different
prices
for
different
type
of
calls
so
you
can
set
you
router
make
the
best
use
of
these
different
tariffs.
Here
are
some
examples
of
some
rules
you
can
set
to
make
best
use
of
each
provider.
Imagine
that
there
are
3
different
VoIP
providers
by
the
name
of
localcheap.com,
Longdischeap.com
and
Mobilecheap.com
1)
Phone
1:
For
Local
calls:
I
use
localcheap.com
who
charges
$0.01
per
minute
to
all
local
calls.
I
set
a
dial
rule,
<:03>[123]x.T
,
on
my
phone
port
1.
1
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Billion 800VGT Router
Localcheap.com
is
the
default
VoIP
provider
I
set
on
phone
port
1.
When
I
dial
any
number
starting
with
1
or
2
or
3
,
,
03
is
always
prepended
in
front
of
these
number.
If
23295
is
dialled,
03-2-32935
is
the actual
phone
number
dialled
using
localcheap.com
as
a
provider.
2)
Phone
1:
For
International
calls:
I
use
longdischeap.com
who
charge
$0.05
per
minute
for
all
International
long
distance
calls.
I
set
a
dial
rule,
0[2456]x.T<@Long_dist_Cheap>
,
on
my
phone
port
1.
Longislcheap.com
is
one
of
the
VoIP
providers
I
set
in
the
User-defined
profile.
No
prefix
is
attached
to
the
dialled
number
when
I
dial
any
number
beginning
with
0
,
followed
by
2,
4,
5
or
6
and
then
the
rest
of
the
phone
number.
If
02
016148513295
is
dialled,
0-2-016148513295
is
the
actual
phone
number
dialled using
longdischeap.com
as
a
provider.
3)
Phone
2:
For
Mobile
calls,
I
use
mobilecheap.com
who
charge
$0.25
per
minute
for
all
local
mobile
calls.
I
set
a
dial
rule,
<123:09>39x.T
,
on
my
phone
port
2.
1
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Billion 800VGT Router
Mobilecheap.com
is
the
default
VoIP
provider
I
set
on
phone
port
2. When
I
call
out
123-39-45678
for
a
mobile
call,
123
is
replaced
with
09
and
09-39-45678
is
the
actual
phone
number
dialled
using
Mobilecheap.com
as
a
provider.
The
Intelligent
Call
Gateway
not
only
saves
time
when
changing
VoIP
settings
to
different
providers
so
as
to
ensure
a
call
gets
routed
using
a
specific
gateway
automatically,
but
it
allows
you
to
take
advantage
of
different
call
rate.
Ring
&
Tone
This
section
allows
advanced
users
to
change
the
parameters
for
the
various
phone
tones
(dial tone,
busy
tone,
answer
tone
and
etc.)The
default
settings
are
the
same
as
the
South
African
PSTN,
but
advanced
users
are
welcome
to
customize
these
values
to
suit
their
requirements.
Country Specific Ring & Tone
Region:
Select
the
country
where
you
are
located
from
the
drop-down
list.
This
VoIP
router
provides
default
parameter
of
ring
tones
according
to
different
countries.
The
ring
-tone
parameters
are
automatically
displayed
after
entering
a
specific
country.
If
your
country
is
not
in
the
list,
or
you
wish
to
have
a
customised
ring
tone,
you
may
manually
create
ring-tone
parameters.
Ring Parameters
Ring
Cadence
(in
ms):
Ring
cadence
is
defined
by
the
following
fields,
Frequency:
On
Time1,
Off
Time1,
On
Time2,
Off
Time2
and
On
Time3,
Off
Time3.
Frequency
is
specified
in
Hertz.
Time
is
given
in
milliseconds.
Tone Parameters
You
may
need
to
check
with
your
local
telephone
service
provider
for
such
information.
Also,
it
is
recommended
that
this
option
be
configured
by
an
advanced
user
only
,
unless
you
are
instructed
to
do
so.
Click
Apply
to
apply
the
settings.
1
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Billion 800VGT Router
Special
Dial
Codes
The
following
table
lists
the
special
dial
codes
that
are
built
into
the
router. Note that Telkom’s VoIP
service may provide the same type of features, and therefore these features might be disabled on the
800VGT platform.
Option
Description
Flash-hook
Switch
to
PSTN
line
Note:
This
is
a
quick
press
of
the
hook
switch.
On
most
phones,
a
button
is
provided
which
provides
Flash-hook
functionality.
The
button
is
marked
"FLASH","RECALL"
or
sometimes
“R”
*69
Return
the
last
missed
call
(for
SIP
service
only)
Note:
Entering
this
on
a
phone
will
call
the
last
number
that
dialled
the
phone.
For
example
A
makes
a
call
to
B,
but
hangs
up
before
B
answers.
If
B
enters
*69,
A
will
be
called
back
.
##
Last
dialled
number
redial
function
*20
Turn
the
Do
Not
Disturb
mode
on
This
mode
enables
a
“Do
Not
Disturb”
feature
on
a
phone,
such
that
any
phone
which
calls
it
will
receive
an
engaged
tone
and
the
called
phone
will
not
ring.
For
example,
B
enters
*20
and
hangs
up.
A
makes
a
call
to
B,
but
receives
the
engaged
tone.
Phone
B
does
not
ring.
*80
Set
the
Do
Not
Disturb
mode
off
*74<x><number>#
Set
the
number
for
Speed
dial
code
‘x’,
where
‘x’
is
a
number
between
2
and
9.
Note:
<x>
is
a
number
between
2
and
9,
and
<number>
is
the
number
to
dial.
To
dial
a
speed
dial
number
from
a
phone
connected
to
the
VoIP
Router
,
dial
:
<
x
>#,
where
<x>
is
a
number
between
2
and
9.
These
settings
will
affect
your
setting
on
the
Speed
Dial
page
of
the
WEB
GUI.
*90<phone-number>
This
is
how
to
make
a
Blind
Call
Transfer
.
With
this
function,
the
specified
<phone-number>
is
the
number
that
you
wish
to
transfer
the
call
to.
This
function
is
for
a
SIP
service
only.
Note:
You
use
Blind
Call
Transfer
when
you
have
a
call
in
progress
(incoming
or
outgoing)
and
decide
you
wish
to
transfer
the
call
to
another
phone.
To
transfer
the
call,
perform
the
following
steps:
1.
Hook-flash
to
get
a
dial
tone.
2.
Dial
*90<phone-number>
(e.g.
*907401).
After
hearing
the
confirmation
tone,
you
can
hang
up.
The
transferred
call
will
hear
a
ringback
tone,
and
the
third-party
phone
(the
one
that
the
call
has
been
forwarded
to)_will
be
rung.
When
the
third-party
phone
is
picked
up,
the
transferred
call
will
be
connected.
If
the
third-party
phone
does
not
answer,
the
caller
being
transferred
must
hang
up
to
cancel
the
connect
attempt.
1
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Billion 800VGT Router
QoS
-
Quality
of
Service
The
QoS
function
helps
you
to
control
your
network
traffic
for
each
LAN
(Ethernet
and/or
Wireless)
application
that
accesses
the
WAN
(Internet).
It
allows
you
to
control
the
quality
and
speed
of
throughput
for
each
application,
when
the
system
is
running
with
a
fully
loaded
upstream
channel.
Here
are
the
items
within
the
QoS
section:
Prioritization
and
Outbound
/
Inbound
IP
Throttling
(bandwidth
management).
Prioritization
There
are
three
priority
settings
provided
in
the
Router:
²
High
²
Normal
(Normal
priority
is
the
default
for
all
traffic
without
any
setting)
²
Low
The
ratio
s of utilization for each priority are
:
High
(60%),
Normal
(30%)
and
Low
(10%).
Click Clear
You can click
Clear
to delete the existing Application.
Application
:
A
user-defined
description
identifying
this
new
policy/application.
Time
Schedule
:
The
details
of
when
this
rule
of
your
prioritization
policy
is
active
.
Priority
:
The
priority
given
to
this
policy/application.
The
default
setting
is
High;
you
may
adjust
this
setting
to
fit
your
requirements.
Protocol
:
The
name
of
the
supported
protocol.
Source
Port
:
The
source
port
of
packets
to
be
monitored.
Destination
Port
:
The
destination
port
of
packets
to
be
monitored.
1
15
Chapter
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Configuration
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