HEARTY

Y WELCOME!
W

TECHNICAL TRAINING

SETU VFXTH
VOIP-FXO-FXS GATEWAY
W Y

Agenda

Overview

Interfaces

Port Configuration

Hardware Architecture

LED Indications

Installation Dos & Don'ts

Applications
li i

Agenda

Programming Using Phone

Programming Using PC

Incoming Call Management

Outgoing Call Management

Advance Settings

F
Features

Maintenance

Status

OVERVIEW
V V W

Overview

A Versatile VoIP-FXO-FXS Gateway

A Gateway that provides voice service over IP network

using SIP protocol

An effective and flexible solution for accessing Internet

based telephone services & corporate Internet systems
across established LAN

Developed to fulfill requirements of SOHO (Small Office

Home Office) users & small/ medium scale enterprises

INTERFACES

Interfaces

USB
(Future Use)

Router
FXO/CO
PC

FXS

Adaptor
p
24V DC
@ 2.5A
60 W

CONFIGURATIONS

VFXTH Configurations
Configurations

VoIP
Channels

FXO
Ports

FXS
Ports

FXO Ports
Label

FXS Ports
Label

VFXTH0016

16

16

P01-P16

VFXTH0024

24

24

P01-P24

VFXTH0032

32

32

P01 P32
P01-P32

VFXTH0800

P01-P08

VFXTH1600

16

16

P01 P16
P01-P16

VFXTH2400

24

24

P01-P24

VFXTH3200

32

32

P01-P32

VFXTH0808

16

P01-P08

P09-P16

VFXTH1212

24

12

12

P01-P12

P13-P24

VFXTH1616

32

16

16

P01-P16

P17-P32

Total 32 SIP Trunks supported in all Configurations

HARDWARE
W
ARCHITECTURE

SLT Modules:
Each Module
supports 2
extensions to
be connected
(Total 8)
Power
Supply
OP V :
+5V
+3.3V,
-27V
-87V

Ethernet Port

SETU VFXTH1616 Hardware

Architecture
TWT
Modules
(
(Total
8)

VoIP module : CODEC IC &

SDRAM. Total 4 such Modules
Each supporting 8 channels

8 BIT
RAM

32 BIT RISC
PROCESSOR
FLASH
RAM
CPLD

Input
Supply : DC
Power Jack
24 V, 2.5A

SETU VFXTH
LED RESET SEQUENCES

LED Indications
Total 34 LEDs in SETU VFXTH1616
Power LED : At Power On Power LED will Turn On
(Continuous Green)
32 Port LEDs : 16 (P01-P16) FXO Port LEDS
16 (P17-P32) FXS Port LEDS
At Initialization:
P01-P32 : OFF
After
f approx 16 sec P01-P03 Glow
l Continuous
i
Redd
After approx 20 sec remaining P04-P32 Glow Red Continuous
After 5 Sec : P01 - P32 LED will be Off

LED Indications
32 FXO/FXS Ports LED indications during normal functioning
C i
Continuously
l Off

P Idl
Port
Idle / Disable
Di bl

400 ms RED on - Incoming Ring Event

200 ms off 400 ms RED on 3000 ms off
( 2 Blinks)
400 ms onOff-Hook Event ((Dialing
g State))
400 ms off
(continuous)
C i
Continuous
On
O
S
Speech
h
Red

System
S
t
LED (STS)

INSTALLATION
DOS &DONTS

Installation Dos

Dust Proof, Moisture Free Location

Away from electromagnetic Sources

Ventilated Location

Path to Static Charges

Stable Mains Supply

Proper Mains Earth

Proper Telecom Earth

Installation Don
Donts
ts

APPLICATIONS

Stand Alone Application

pp
2001
FXS1FXS16
2016
FXO1FXO16

Ethernet

StandAloneCallPossibilities

PSTN
Network

[Link] IPNetwork
2. FXS PSTNNetwork
[Link]
PSTN Network
Broadband
Modem/Router

IP
IP
Network

In Front of PBX Application

pp
SETU
VFXT1616

PBX

FXS1FXS16

FXO1FXO16
FXS1FXSN

FXSportsofVFXTHare
connectedtoFXOPortsof
[Link]
canavailthePSTN&VoIP
NetworksofVFXTH

Ethernet

Broadband
Modem/Router

PSTN
Network

IP
Network

Behind the PBX Application

pp
SETU
VFXT1616

2001
FXS1FXS16

2016

FXOportsofVFXTH
areconnectedtoFXS
PortsofPBX.
ExtensionsofVFXTH
thus can use the
thuscanusethe
TrunkofVFXTH

PBX
PSTN
N/W

FXO1FXO16

Ethernet

FXS1FXSN

Broadband
Modem/Router

IP
Network

Analog Extension PBX Over IP Application

SETU
VFXT1616

SETU
VFXT1616

2001
FXS1
FXS16

2001
FXO1
FXO16

2016

Broadband
Modem/Router
PSTN
N/W

FXO1
FXO16

Ethernet

FXO1
FXO16

IP
Network

Ethernet

Broadband
Modem/Router

PBX

FXS1FXSN

Peer to Peer Callingg

SETU
VFXT1616

2001

IP
Network

SETU
VFXT1616

2001
FXS1
FXS16

FXS1
FXS16

FXO1
FXO16

FXO1
FXO16

Ethernet

Ethernet

2016

PSTN Call over IP (Long Distance

converted to Local Call)
SETU
VFXT1616

2001

IP
Network

SETU
VFXT1616

2001
FXS1
FXS16

FXS1
FXS16

FXO1
FXO16

FXO1
FXO16

Ethernet

Ethernet

Mumbai

Delhi

2016

PROGRAMMING USING PHONE

Programming Using Phone

Certain parameters of SETU VFXTH can be

configured by dialing system commands from a
t l h
telephone
connected
t d to
t the
th FXS portt

You can configure

g
certain network parameters
p
like
IP address, Subnet Mask, Connection Type, set the
system to default
d f l andd also
l view
i current IP address,
dd
Subnet Mask, Connection Type, DNS and Gateway
address by dialing system commands

SE Login
Connect Analog Phone to FXS port of SETU VFXTH
OFF - Hook the phone
Hear Dial Tone [Toooooooooooooooooooooo]
Dial Command #19
#19 SE Password
Password for Login
Default SE Password is 1234
Enter System Commands to perform different functions
Dial 00#* to Exit from Programming mode

Commands

11 IP Address #* (To change IP Address)

12 Subnet Mask #* (To change Subnet Mask)

10 Code #* (to change the connection type)

[1 static, 2 DHCP, 3 PPPoE]

31 Code ##* (To Enable1/Disable0 VLAN Tag)

51 Reverse SE Password #* (To Restore

Factory defaults)

Commands

21 #* ((To view IP Address)) & go

g On Hook

22 #* (To view Subnet Mask) & go On Hook

23 #* (To view Gateway Address) & go On Hook

24 #* (To view DNS Address) & go On Hook

20 #
#* (to view the connection type) & go On hook

27 SIP Trunk Number (1 9) #* & go On hook

(To view the status of SIP Trunk)

PROGRAMMING USING PC

SETU VFXTH : Configuration

Web Jeeves Login from Local Network
NetworkSwitch
[Link]
SETUVFXTHislocated
onLocalIP

[Link]

SETU VFXTH : Configuration

Web Jeeves Login from Public Network
Internet

[Link]
SETUVFXTHislocated
on Global IP
onGlobalIP

PCwithinternet
connection

SETU VFXTH: Configuration

Web Jeeves Login from Public Network
LAN:
[Link]
Internet
WAN:
203 88 123 231:80
[Link]:80

Routersport:80is

80 i
forwardedtoIPAddressof
SETUVFXTH
PCwithinternet
connection

IP:[Link]
: 9 . 68. . 5
Subnet:[Link]
Gateway:[Link]

Programming
Built
B ilt in
i Web
W b server
GUI based software called JEEVES
Accessible using any web browser
Default IP of Ethernet Port is [Link]
Default SE password is 1234

Click on Start Internet

Explorer (Any Browser)

Programming

Enter Ethernet Port IP

Address of SETU VFXTH

Login Page

Enter SE Password for

Login (Default: 1234)

Home Page

Network Port Parameters

This parameters can be programmed as per existing data network

Connection type :

1 Static:
1.
S i IP address,
dd
S
Subnet
b mask
k & Gateway
G
Add
Address assigned
i d
Manually
2. DHCP: IP address, Subnet mask & Gateway Address assigned
automatically by DHCP server
3. PPPoE: Select this option if your ISP provides internet services
using PPPoE, If you select this option you must enter the User
ID, password and service name in PPPoE parameters

Network Port Parameters

Select connection type of SETU
VFXTH and according to the
connection type program the IP details

INCOMING CALL MANAGEMENT

- SIP TRUNK
- FXO PORT

Incoming Call Route

The process of routing calls originated on FXS Port,
Port
FXO port and SIP trunks to the destination port in
SETU VFXTH takes place in two steps:
1. Determination of destination number
2. Determination of destination port

DESTINATION NUMBER
DETERMINATION

FXO PORT

Destination Number Determination on FXO Port

Incoming Call Route

options on FXO Port

Destination Number Determination on FXO Port

Without
ih
any Destination
i i Number
b
To a Fixed Destination Number
On the basis of Callingg Party
y Number
After answering the call and collecting the digits

Without Any Destination Number

Incomingg call on the FXO pport
All calls received on the FXO port are directly
routed to the fixed destination port, configured for
this pport,, regardless
g
of the destination number

Without Any Destination Number

FXO
FXS

022 2631725

2001

SETU VFXTH

Without Anyy Destination Number

2 different routings
defined here
1. Route all IC
calls (with
CLI)
2. Route all IC
calls (without
(
CLI)

Select route for all

Incoming calls as Without
any Destination
D i i number
b

Route To a Fixed Destination Number

Incoming call on the FXO port
Call is routed to the Fixed destination number
programmed on that particular trunk line using the
Destination pport programmed
p g
for that trunk
Destination port can be FXS port, FXO port or SIP
Trunk

Route To a Fixed Destination Number

FXO
SIP

471@[Link]
b d
li

0265 2630555

SETU VFXTH

Fixed Destination Number: 471

Route To a Fixed Destination Number

2 different routings
defined here
1. Route all IC calls
(with CLI)
2. Route all IC calls
(without CLI)

Define fixed
d ti ti number
destination
b
on which you want
to route the call

Define destination
port for routing calls

Enable this flag if you

want to block the IC
calls received without
CLI on this FXO port

Route on the basis of Callingg Partyy Number

Incomingg call on the FXO pport
Calls are routed to a specific number according to
the calling party number
When there is an incoming call on the FXO port,
port
SETU VFXTH will match the calling party number
with the entries of the calling party number based
table, if a match is found, the call is routed to the
destination number

Route on the basis of Callingg Partyy Number

FXO
SIP

471@[Link]

0265 2630555

SETU VFXTH
CallingNumber

Destination Number

02652630555

471

02226471110

472

Route on the basis of Calling Party Number

2 different routings
defined here
1. Route all IC
calls
ll (with
( ith
CLI)
2. Route all IC
calls (without
CLI)

Select route for all

Incoming calls as on
the basis of calling
party number

Route on the basis of Callingg Partyy Number

Program
g
the callingg
number based table with
calling party number
and destination number

After Answeringg the call & collectingg the digits

g
Incoming
I
i call
ll on the
th FXO portt
Incomingg calls are answered and dial tone is pplayed
y
to the caller, allowing the caller to dial the desired
number
b
The number dialed byy the caller is considered as the
destination number and dial it out using the
d ti ti portt programmedd
destination

After Answeringg the call & collectingg the digits

g
471
Dial Tone

FXO
SIP

0265 2630555
471@[Link]

SETU VFXTH

After Answering the call & collecting the digits

Select route for all Incoming

calls as after answering the
call and collecting the digits

2 different routings
defined here
1 Route all IC
1.
calls (with CLI)
2. Route all IC
calls (without
CLI)

Define destination
port for routing calls

SIP TRUNK

Destination Number Determination on SIP Trunk

Incoming Call Route

options on SIP trunk

Destination Number Determination on SIP Trunk

Without
ih
any Destination
i i Number
b
To a Fixed Destination Number
On the basis of Callingg Party
y Number
After answering the call and collecting the digits

Without Any Destination Number

Incomingg call on the SIP Trunk
All calls received on the SIP Trunk are directly
routed to the fixed destination port, configured for
this pport,, regardless
g
of the destination number

Without Any Destination Number

SIP
FXS

022 2631725

2001

SETU VFXTH

Without Anyy Destination Number

2 different routings
defined here
1. Route all IC
calls (with
CLI)
2. Route all IC
calls (without
(
CLI)

Select route for all

Incoming calls as Without
any Destination number

Define destination
port for routing calls

Route To a Fixed Destination Number

Incoming call on the SIP Trunk
Calls are routed to the Fixed destination number
programmed on that SIP trunk using the Destination
pport programmed
p g
for that SIP trunk
Destination port can be FXS port, FXO port or SIP
Trunk

Route To a Fixed Destination Number

SIP
FXO

0265 2630555

471@[Link]

SETU VFXTH

Fixed Destination Number: 0265 2630555

Route To a Fixed Destination Number

Enable this flag if you want to

block the IC calls received
without CLI on this SIP Trunk
2 different routings
g
defined here
1. Route all IC calls
(with CLI)
2. Route all IC calls
(without CLI)

Define fixed
d ti ti number
destination
b
on which you want
to route the call

Define destination
port for routing calls

Route on the basis of Callingg Partyy Number

Incomingg call on the SIP Trunk
Calls are routed to a specific number according to
the calling party number
When there is an incoming call on the SIP trunk,
trunk
SETU VFXTH will match the calling party number
with the entries of the calling party number based
table, if a match is found, the call is routed to the
destination number

Route on the basis of Callingg Partyy Number

SIP
FXO

0265 2630555

471@[Link]

SETU VFXTH
CallingNumber

Destination Number

02652630555

471

02226471110

472

Route on the basis of Calling Party Number

Select route for all

Incoming calls as on
the basis of calling
party number
2 different routings
defined here
1. Route all IC
calls (with CLI)
2. Route all IC
calls (without
CLI)

Define destination
port for routing calls

Route on the basis of Callingg Partyy Number

Program the calling

number based table with
calling party number
and destination number

To the Called Partyy Number

Incoming
I
i call
ll on the
th SIP Trunk
T k
Incomingg calls are routed to a desired number
depending upon the called number received in the
SIP ID off requestt URI off th
the INVITE message

To the Called Partyy Number

0265 2630555

SIP
FXO

02652630555@
[Link]

0265 2630555

SETU VFXTH
[Link]

To the Called Partyy Number

2 different routings
defined here
1 Route all IC
1.
calls (with CLI)
2. Route all IC
calls (without
CLI)

Select route for all

Incoming calls as to the
called party number

Define destination
port for routing calls

DESTINATION PORT
DETERMINATION

Destination Port Determination

SETU VFXTH supports different methods of
determining the destination port for the calls
originated on FXS Port, FXO Port and SIP trunks,
they are:
1. Fixed
2 On the basis of destination number
2.
3. On the basis of calling party number (Not
Supported on FXS Port)

Destination Port Determination on FXS Port

Destination port determination

options on FXS port

Destination Port Determination on FXO Port

Destination port determination

options on FXO port

Destination Port Determination on SIP Trunk

Destination p
port determination
options on SIP Trunk

DESTINATION PORT
DETERMINATION FIXED

Fixed

Click on Edit to
change the members
of routing group

Program the routing group

for routing of Incoming
calls on the SIP trunk

Fixed
Program CLI
P
number to be sent on
destination port
Program Group
Member for
routing group

Click here to Apply

fallback routing
group

Fallback Routing group is used in case if all members of

routing group are busy or the trunk line is down

CLI display on Destination Port

2001

3001

[Link]

ReceivedCalledParty
3001
ReceivedCallingParty
2001

2002
2003

2001
3001

IP
Network

3001
3002
3003

SETUVFXTH
SETU
VFXTH
INDIA

[Link]

SETUVFXTH
USA

DESTINATION PORT
DETERMINATION ON THE
BASIS OF DESTINATION NUMBER

On the Basis Of Destination Number

Program the Destination

number and routing group
in the destination number
based routing table

Click on Add to add

the
h new entry ffor
destination number

Click on Delete to
d l t th
delete
the selected
l t d entry
t
for destination number

On the Basis Of Destination Number

Enter the destination
number for which routing
group is to be programmed

Program the routing group

and fallback routing group
for the destination number
defined above

DESTINATION PORT
DETERMINATION ON THE BASIS
OF CALLING PARTY NUMBER

On the Basis Of Callingg Partyy Number

Program the calling

number and routing group
in the calling number
based routing table

Click on Add to add

the new entryy for
Calling Number

Click on Delete to
delete the selected entryy
for Calling Number

On the Basis Of Callingg Partyy Number

Enter the Calling number
f which
for
hi h the
h routing
i group
is to be programmed

Program the routing

group and fallback
routing group for the
Calling number
defined above

OUTGOING CALL MANAGEMENT

- FXS PORT
- FXO PORT
- SIP TRUNK

Call Block on FXS Port

For OG Call we can

allow or block outgoing
calls, enable flag to
Block the Outgoing from
this trunk

Call Block FXO Port

For OG Call we can

allow or block outgoing
calls,
ll enable
bl flag
fl to
t
Block the Outgoing from
this trunk

Call Block on SIP Trunk

For OG Call we can

allow or block outgoing
calls,
ll enable
bl flag
fl to
t
Block the Outgoing from
this trunk

STUN

STUN ((Simple
p Traversal of UDPs
through NATs)

When the VoIP port (WAN) is located behind a

NAT Router & SIP Messages need to forwarded to
the Public Internet

STUN specifies the mechanism required for NAT

t
traversal
l in
i SIP messages. STUN server facilitates
f ilit t
traversing through most NATs except symmetric
NATs

Illustration of STUN

STUNRequest

STUNRequest
Source:[Link]:5060

Source: [Link]:5060
Source:[Link]:5060

STUNResponse
To:[Link]:5060
Payload:[Link]:5060

STUNServer

STUNResponse
To:[Link]:5060
Payload:[Link]:5060

STUN

Program the STUN Server Address; Listening Port of

STUN Server (1024-65535) Default port : 03478; Enable
th Flag
the
Fl Use
U SIP P
Portt ffetched
t h d using
i STUN if SIP portt
required to be fetched by STUN else disable when Port
Forwarding in the Router is done for SIP messages

STUN

Select NAT type as STUN if you want

to use IP address
dd
fetched
f h d using
i STUN

STUN

Status page will display the IP

address port nnumber
address,
mber and NAT
type fetched using STUN

ROUTER PUBLIC IP ADDRESS

Routers Public IP Address

Port Forwarding :
Since STUN doesnt work with symmetric NAT , as an
alternative
lt
ti to
t STUN Port
P t Forwarding
F
di can be
b done
d
i the
in
th
router and Routers Public address that is configured
can be used as Source Port IP Address

Router Public IP Address

Use NAT type as Router

Public IP address

Router Public IP Address

Program Router Public

IP Address here

Router Public IP Address

Status page will display

the Router Public IP
address programmed in
th system
the
t parameter
t page

P2P Call One Device is on Public IP and

Oth D
Other
Device
i iinstalled
t ll d behind
b hi d NAT

Port Forward in
Router

Router separates
Private and Public
Network

Internet
LAN
[Link]

WAN
[Link]

SETU VFXTH
IP: [Link]
G/W : [Link]

Private IP

[Link]

Public IP

Router Configuration: Example

Routers
Network
Parameters

*LinksysisawhollyownedsubsidiaryofCiscoSystems,Inc.

Router Configuration: Example

Port Forwarding:
Routers SIP an RTP Port
forwarded to Private IP of
SETU VFXTH

*LinksysisawhollyownedsubsidiaryofCiscoSystems,Inc.

ADVANCE SETTINGS

Access Codes

Access code is a string of digits dialed to use a feature

SETU VFXTH users can access the features and facilities by

dialingg the access code assigned
g
to them from a pphone.
SE/User can

1 Enable/Disable a feature
1.
2. Access Supplementary feature
3. Enter into the programming mode

SETU VFXTH provides default access code for all features,

features
you can change it to suit your preferences

Access Codes

Access codes can be

changed from here

Access Codes

Access codes can be

changed from here

Allowed Denied Numbers

This feature provides the flexibility to allow or deny dialing of

a particular number or a set of numbers from a particular port
p
or all ports

Allowed Denied number logic makes use of two number lists:

1. Allowed Numbers List: this is the list of numbers that can be

dialed out from the SIP trunk (default number list 7)
2. Denied Numbers List: this is the list of numbers that are to be
restricted from being dialed out from the SIP trunk (default
number list 8)

Allowed Denied Logic on FXS Port

Apply allowed denied list on FXS

Port & program the number list
for allowed & denied numbers

Allowed Denied Logic on FXO Port

Apply allowed denied list on FXO

pport & program
p g
the number list for
allowed & denied numbers

Allowed Denied Logic on SIP Trunk

Apply allowed denied list on SIP

trunk & program the number list for
allowed & denied numbers

Automatic Number Translation

This feature is used to translate the dialed number string to

preprogrammed number string

ANT can be used to modify, add or delete the prefix of the

destination number string

For this feature we need to configure dialed number string

and substitute number string in number list table

ANT feature is applied on destination ports (On all SIP

trunks and FXO Ports)

Automatic Number Translation

Apply ANT on FXO Port and

program the number list for dialed
aandd substitute
subst tute number
u be string
st g

Automatic Number Translation

Apply ANT on SIP Trunk and

program the number list for dialed
and substitute number string

Automatic Number Translation

Program
g
dialed and substitute number
strings in the number list table

Black Listed Callers

SETU VFXTH supports feature Black

Black listed Callers
Callers which
enables you to block incoming calls from specific numbers
andd addresses
dd
on the
th SIP ttrunks
k

This feature is applicable on source port only

To use this feature, user must configure the numbers of

unwanted
n anted callers in a number
n mber list

Enable the Reject Calls from Blacklisted Caller check box

on the SIP trunks on which you want to apply this feature

Black Listed Callers

Apply black listed caller

feature on selected SIP
trunk and define the
number list for the same
Black Listed Callers

Black Listed Callers

Program
g
the number list with
the CLI of black listed callers

Call Detail Record (CDR)

Itss a record for the calls,

It
calls containing information about the
gateways usage when call was made

Maximum of 2000 call record entries can be stored

Call record entries are stored in FIFO logic

User can set different filters as required and generate Call

Detail Record (CDR) report

Call records can be cleared manually at any time

Call Detail Record (CDR)

1.
2.
3
3.
4.
5.
6.
7.
8.
9.
10
10.
11.

It is possible to get following details of a call with CDR

Date of call origination
Time of call origination
Calling number
n mber
Called number
Duration of call
Source port
Destination port
Disconnected by
Cause
PIN number
b
Remarks

Call Detail Record (CDR)

Below mentioned filter can be programmed for CDR

1. The port from which the calls originate (Source Port)

2. The port on which the calls terminate (Destination Port)
3. Calls made on particular dates
particular time
4. Calls made at a p
5. Calls of a certain duration
6. Calls of certain called party numbers
7. Calls of certain calling party numbers
8 Calls made with PIN authentication
8.
9. Calls made without PIN authentication

Call Detail Record (CDR)

Set filter parameters
for CDR here

Click here to clear

all call records

Call Detail Record (CDR)

Click on download to get

Zip file containing CDR
in .csv and .txt format
Save Zip file & extract it to
get CDR in .csv and .txt file

Call Detail Record (CDR)

CDR can also be

viewed from JEEVES

Call Detail Record (CDR)

CDR can also be

viewed from JEEVES

PIN Authentication

PIN authentication is a security feature to restrict access to the

system and prevent possible misuse of resources

User can use the PIN authentication on the source port to establish
identity of callers before their call is processed by SETU VFXTH

PIN authentication can be used on the source port only if the

incoming call routing for the source port is set to After answering
the call and collecting digits

To use this feature it must be enabled on the source port and the
PIN authentication table must be configured

PIN Authentication

The PIN authentication table stores up to 500 PIN numbers and

their corresponding authentication passwords

If PIN
N au
authentication
e ca o iss enabled
e ab ed on
o source
sou ce port,
po , SETU
S U VFXTH
V
answers the Incoming call and plays a feature tone, it waits for the
caller to dial the PIN number and ppassword,, it matches them with
the PIN authentication table, if match is found it allows the call to
be processed

In case of wrong PIN entered, SETU VFXTH allows the caller to

p , if the caller fails to dial correct PIN and
make two more attempts,
password in all attempts, the system disconnects the call

PIN Authentication FXO Port

Select routing type

f answering
after
i the
h
call and collecting
the digits for PIN
authentication
feature to use

Enable
E
bl this
thi flag
fl
for prompting
caller to enter PIN

PIN Authentication

Enter PIN number & PIN password,

system checks PIN entered by the caller
duringg call with the entries in the PIN
authentication table, if match found then
only the call will be processed further

Peer to Peer Dialing

Making an IP call without the intervention of a proxy server

is called peer to peer calling

As peer to peer calling does not require a proxy server, voice

communication using this application can be done virtually
free of cost

The major cost savings offered by this application makes it a

very attractive mode of inter branch or intra office voice
communication

Peer to Peer Callingg

SETUVFXTH
Location A

2001

IP
Network

SETUVFXTH
Location B

3001
FXS1
FXS16

FXS1
FXS16

FXO1
FXO16

FXO1
FXO16

Ethernet

[Link]

Ethernet

[Link]

3XXX

Peer to Peer Dialing

Program the peer to peer table

with destination number &
destination address (IP address
of opposite location)
Click here to add new
entry to the table

Click
li k here
h to delete
d l
entry from the table

Digest Authentication

Digest authentication is a challenge based authentication service

of SIP to authenticate the identity of the originator of SIP request in
the INVITE message

The recipient of the request can ascertain whether or not the

originator of the request is authorized to make the request

When the digest credentials of the originator User Name and

g are authenticated and accepted
p
Password in the INVITE message
by the recipient, the originator and recipient are connected

You
ou may
ay use tthee digest
d gest authentication
aut e t cat o to restrict
est ct access to SETU
S U
VFXTH to specific callers, prevent unwanted or malicious calls

Digest Authentication

When this feature is enabled on a SIP trunk for all Incoming calls

1. SETU VFXTH will challenge the identity of the calling party

2 When the calling party sends its credentials
2.
credentials, SETU VFXTH
authenticates the credentials by matching it with its Digest
A h i i table
Authentication
bl
3. If a match is found, the calling party will be authenticated and the
call will be allowed on the SIP trunk
4. If no match is found, SETU VFXTH will consider it as invalid
authentication information and reject the call

Digest Authentication

Enable apply flag in

SIP trunk to use
digest authentication

Digest Authentication

Enter Digest credentials (User ID

and User Password) of calling party

Static Routing
Static Routingg Table is required
q
when you
y have more than one
router (Gateway) in your network and you want SETU
VFXTH to send packets to multiple routers/gateways for
different types of calls
If you have only one router connected in the network , you
need not configure this table & LAN interface of router will
act as the default gateway for the system

Static Routing

Program the static routing table with

the details, if the match is found
here then gateway will send the
packets to defined gateway address
opposite to the destination address

Prefix to Domain Name Conversion

Prefix to domain name conversion is used when a user sets

call forward or makes a blind transfer on SIP, this feature is
applicable only when the destination port is SIP

SETU VFXTH supports multiple SIP trunks & FXS ports,

when a FXS port user dials a SIP number
number, SETU VFXTH
routes the call to the IP network using the SIP trunk
determined by the routing mechanism
mechanism. The FXS user can dial
only numbers not domain names, therefore it becomes
necessary that the domain names be assigned prefix codes
which the FXS user can dial

Prefix to Domain Name Conversion

User need to program prefix v/s domain name in the table

This table is not checked for making an outgoing call, but it

is checked when some FXS port has set call forward and
onlyy number is pprogrammed
g
or user is doing
g blind transfer

For example prefix in the table is programmed as *123 and

d
domain
i name as [Link]
b
andd destination
d i i number
b for
f call
ll
forward is *1239974 then it will be replaced by
9974@[Link]

Prefix to Domain Name Conversion

Define prefix and domain

name in the table

Disconnect Tone

If call disconnection is signaled by your CO network in the

form of disconnect tone on the FXO Ports

You must enable Disconnect Tone Detection on the FXO

port and select the Disconnect tone type

To enable the system to detect the disconnect tone accurately,

you must configure the cadence and frequency of the
disconnect tone type you selected, as supported by the CO
network

Disconnect Tone

Enable disconnection
tone detection here

Disconnect Tone

Program the disconnect

tone cadence here

Emergency Numbers
SETU VFXTH supports
pp
dialingg of emergency
g y numbers from
all ports, Emergency numbers and their respective routing
groups must be configured in the emergency number table
User can configure up to 10 numbers of emergency services
such as ambulance, fire brigade, police etc.
By default
default, No emergency numbers are loaded in the system,
system
in the emergency number table

Emergency Numbers

Click here to Edit

entry of the table

Click here to add new

entry to the table

Click here to delete

entry from the table

FEATURES

Class Of Service

If any FXS port want to use supplementary services then these

services must be activated in COS for particular FXS port as well as
at SIP services provider in case of SIP account calling

SETU VFXTH offers following telephony features, which they can

access by dialing
d a g access codes

1. Call Hold

6. Blind Transfer

2. Call Forward

7. Attended Transfer

3. Call toggle

8. Do Not Disturb (DND)

4. Call waiting

9. Hotline

5. Conference

Class Of Service

Enable required
feature from Class
of service on
particular FXS port

Supplementary Services

Enable the
supplementary
services after
enabling the
feature in COS

Subscriber Type

Wh SETU VFXTH is
When
i interfaced
i t f d with
ith service
i provider
id
server ITSP or other PBX that supports supplementary
services that require dialing of Flash like call hold, call
transfer, call waiting, you must select the subscriber type
according to the extent of feature access you want on the FXS
pport connected to the system
y

Subscriber Type

Select Network if you want to use supplementary services

supported by the other PBX, you can access the service
provider
id features
f t
by
b dialing
di li FLASH,
FLASH you will
ill nott be
b able
bl to
t
access the local features of SETU VFXTH

Select Gateway if you want to use supplementary services

supported
suppo
ed by thee S
SETU
UV
VFXTH,, in thee ggateway
ew y mode
ode you
will also be able to access the supplementary services of the
service provider which require dialing of FLASH

Subscriber Type

Select the
subscriber type
of your choice

FXS Port
Signaling
g
g

Loopp Start

Connector

RJ-45

Off-Hook Line Impedance

p

600 / 900 / Complex

p

No. of Long Loop Extension

Loop Limit

1800 (Max) Excluding Telephone

Set

On-Hook Voltage (Tip/Ring)

-48 V

Off-Hook Current

25 mA (Max)

Ringing Voltage

Trapezoidal 60 VRMS/25Hz and

Sinusoidal 52VRMS/25Hz

FXS Port
REN

DTMF D
Detection
t ti

ITU T Q.24
ITU-T
Q 24

CLI Presentation

DTMF, FSK ITU-V23 & FSK Bellcore

Protection

Over Voltage Secondary Protection

Return Loss

>18 dB

Longitudinal Balance

>50 dB

Transmission Level
Adjust
j
Answer Signaling on
FXS
Disconnect Signaling
on FXS

Tx Gain : -3dB to +6dB ;

Rx Gain : -3dB to +6dB
Battery Reversal
Battery Reversal & Open Loop
Disconnect

FXS Port

Hardware settings
on FXS port

FXS Port

General settings
on FXS port

FXS Port

General settings
on FXS port

FXS Port

First Digit & Inter

Digit wait timer

FXS
First Digit Wait Timer:
Signifies the time for which the system waits for
receiving a first digit after going off-hook from FXS
port
On expiry of this timer, system will give error tone to
the user
It is programmable from 01 to 99 seconds (Default: 15
seconds)

FXS
Inter Digit Wait Timer:
Signifies the time period between 2 consecutive digits
while the system is receiving the digits from caller
On expiry of this timer, ATA1S will process the digits
dialed so far by the user
it is programmable from 01 to 99 seconds (Default: 5
seconds)

FXO Port
Return Loss

>18 dB

Longitudinal Balance

>50 dB

T
Transmission
i i Level
L l Adjust
Adj

Tx Gain:
T
G i -15
15 dB to +10
10 dB
Rx Gain: -15 dB to +10 dB

Call Maturity

Delay & Polarity Reversal

Answer Supervision on FXO

Battery Reversal

Disconnect Supervision on
FXO

Battery Reversal & Open Loop

Disconnect

FXO Port
REN

DTMF Detection
D
i

ITU T Q.24
ITU-T
Q 24

CLI Presentation

DTMF, FSK ITU-V23 & FSK Bellcore

P t ti
Protection

O
Over
V
Voltage
lt
Secondary
S
d
P
Protection
t ti

Return Loss

>18 dB

Longit dinal Balance

Longitudinal

>50 dB

Transmission Level
Adjust
Answer Signaling on
FXS
Disconnect Signaling
on FXS

Tx Gain : -3dB to +6dB ;

Rx Gain : -3dB
3dB to +6dB
6dB
Battery Reversal
Battery Reversal & Open Loop
Disconnect

FXO Port

Hardware settings
on FXO port

FXO Port

General settings
on FXO port

Making a new call using access code

This feature enables callers to disconnect the current call and make a
new call
ll using
i SETU VFXTH without
ih
getting
i disconnected
di
d from
f
the system

This feature is useful when you want to make multiple calls without
getting disconnected each time their call ends

This feature is applicable only on the FXO port and only when
After answering the call and collecting digits is selected as the
d i i number
destination
b determination
d
i i method
h d

If you have enabled Connect source port when number is out

dialed on the FXO port, you will not be able to provide this feature
to callers

Making a new call using access code

To make a new call using access code

In speech with the current call

Dial
Di l #91
Current call will disconnect
Dial the new number you want to call
Speech will be establish on the new call as called party
answers the call
While in speech
p
dial #91 again
g to make another new call

Making a new call using access code

Enable the flag to allow

user making new call
using access code

Disconnecting a call using access code

SETU VFXTH enables user to disconnect a call using an access code

When the call disconnect access code is dialed, SETU VFXTH

g g in the call
releases the pport engaged

This feature is applicable only when destination number

determination method is selected as After answering the call and
collecting digits

If you have enabled Connect source port when number is out

dialed on the FXO port or have enabled Connect source port when
183 is received on SIP on the SIP trunk, you will not be able to
provide this feature to users

Disconnecting a call using access code

Enable the flag to allow

call disconnection using
access code

Disconnecting a call using access code

Enable the flag to allow

call disconnection using
access code

IP Dialing

SETU VFXTH supports direct dialing of IP addresses from the source

port. To provide IP dialing facility to the users, you must configure a
SIP trunk or a SIP group for IP dialing

IP number can be dialed with dot . as entered by * while dialing it

For e.g. to dial IP address [Link] dial as 192*167*100*1

192 167 100 1 from
the Phone at FXS

When an IP address is dialed from the source port of SETU VFXTH,

VFXTH
the system does not check the destination port determination method
you have configured for that port,
port instead it routes the dialed IP address
through the SIP trunk or SIP group you configured for IP dialing

IP Dialing

SIP trunk or SIP

trunk group can be
defined IP dialing

SIP Timers
SIP Invite Timer
SIP Provisional Timer
General Request Timer

SIP Invite Timer

It is the time for which VFXTH waits for a response

from the called party after sending INVITE message

This time starts after sending INVITE message to the

called ppartyy and stops
p on receipt
p of pprovisional
response or final response or when the user goes ONH k on expiry
Hook,
i off the
h timer
i
the
h call
ll is
i disconnected
di
d

g of SIP INVITE Timer is 10 - 80 seconds

The range
(Default: 30 Seconds)

SIP Provisional Timer

It is the time for which VFXTH waits for final response

after receiving provisional response from the called party

This timer starts on receipt of provisional response from

the called party and stops on receipt of final response from
the called party or when the user goes ON-Hook, on expiry
of the timer the call is disconnected

The range
g of SIP Provisional Timer is 10 - 180 seconds
(Default: 60 Seconds)

General Request
q
Timer

It is the time for which VFXTH waits for the response

p
of a transaction request

This timer starts on initiating a transaction

This timer stops on receipt of a response for the request

On expiry of timer, the VFXTH clears the transaction

The range of SIP Provisional Timer is 10 - 60 seconds

(Default: 20 Seconds)

SIP Over TCP

The SIP over TCP option allows you to
send/receive the SIP messages over TCP
SIP over TCP is applicable for both Proxy and Peerto Peer
to-Peer
By Default SIP messages transported over TCP
Disable the flag to send SIP messages over UDP

System
y
Parameters

Program the timer

values according
to the requirement

MAINTENANCE

Firmware/Configuration
Browse the ZIP file with file name
[Link] having configuration
files & click on Restore

Click on save to get the

configuration backup

Browse the ZIP file with file name

[Link] having new
firmware files & click on Upgrade
to upgrade
d the
h system fi
firmware

Click on download to get

Zip
i file
fil containing
i i CDR
in .csv and .txt format

System Debug

Debugs are logs of actions and events that take place on system,
these logs are useful for troubleshooting and system security

SETU VFXTH supports Syslog client for debugging,

debugging Syslog
client enables the system to send debug messages in Syslog
f
format
t to
t the
th remote
t Syslog
S l server on the
th IP network
t
k

Syslog uses the UDP as transport protocol

To be able to use this feature, you must enable Syslog,

configure
g
the Syslog
y g Server Address and define the server pport
on which the Syslog will listen for debug messages

System Debug

Debug events
can be viewed
on the screen

System Debug

Program the IP address and

port number of PC/Laptop
where Syslog server is installed
Debug
D
b ffor P
Port:
t clear
l the
th check
h k
box to disable the debug for the
port which is not needed

System Debug

Click on Save Debugg to save the

logs captured into the system

PCAP Trace

PCAP or Packet capture consists of intercepting and logging the

traffic passing over the network, PCAP intercepts each packet in the
data streams that flow across the network, and can decode and analyze
its contents

A maximum 2MB of packets can be captured and stored in the system

SETU VFXTH also supports filters and promiscuous mode for

capturing packets

If promiscuous mode is enabled, SETU VFXTH will capture all

network traffic and if disabled then system will capture only traffic
that is directly related to SETU VFXTH (to or from SETU VFXTH)

PCAP Trace

Click here to Enable

Promiscuous mode

Click here to start

the PCAP trace

Click here to stop

the PCAP trace

Enter the filter

details here

Once the PCAP is

captured save the trace
file on your PC/Laptop

Default System

Click OK to factory
factor
default the gateway

Soft Restart

Click OK to Restart
SETU VFXTH

STATUS

Network

Network status
with IP details

FXO Port

FXO Port status whether

Line is connected or not

SIP Trunk

SIP trunk status

disabled, active etc.

Reason off ffailure

R
il
in
i
case Registration failed

Firmware

Firmware Version
Revision display

MATRIX COMSEC PVT. LTD.

|Vadodara | Gujarat | India |
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