Matrix Voip User Manual

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
1.FXS IPNetwork
2. FXS PSTNNetwork
2.FXS
PSTN Network
Broadband
Modem/Router
IP
IP
Network
In Front of PBX Application

pp
SETU
VFXT1616
PBX
FXS1FXS16
FXO1FXO16
FXS1FXSN
FXSportsofVFXTHare
connectedtoFXOPortsof
PBX.ExtensionsofPBXthus
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
192.168.50.200
SETUVFXTHislocated
onLocalIP
192.168.50.33
SETU VFXTH : Configuration

Web Jeeves Login from Public Network
Internet
203.88.123.231
SETUVFXTHislocated
on Global IP
onGlobalIP
PCwithinternet
connection
SETU VFXTH: Configuration

Web Jeeves Login from Public Network
LAN:
192.168.1.1
Internet
WAN:
203 88 123 231:80
203.88.123.231:80
Routersport:80is
80 i
forwardedtoIPAddressof
SETUVFXTH
PCwithinternet
connection
IP:192.168.1.151
: 9 . 68. . 5
Subnet:255.255.255.0
Gateway:192.168.1.1
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 192.168.001.147
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
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
FXO
SIP
471@matrixpbx.dynalias.org
b d
li
0265 2630555
SETU VFXTH
Fixed Destination Number: 471
defined here
1. Route all IC calls
(with CLI)
(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
FXO
SIP
0265 2630555
SETU VFXTH
CallingNumber
Destination Number
02652630555
471
02226471110
472
Route on the basis of Calling Party Number
defined here
1. Route all IC
calls
ll (with
( ith
CLI)
2. Route all IC
calls (without
CLI)

Incoming calls as on
the basis of calling
party 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
SETU VFXTH
After Answering the call & collecting the digits
Select route for all Incoming

calls as after answering the
call and collecting the digits
defined here
1 Route all IC
1.
calls (with CLI)
2. Route all IC
calls (without
CLI)
Define destination
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

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
SIP
FXS
022 2631725
2001
SETU VFXTH
Without Anyy Destination Number
defined here
1. Route all IC
calls (with
CLI)
2. Route all IC
calls (without
(
CLI)

Incoming calls as Without
any Destination number
Define destination

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
SIP
FXO
0265 2630555
SETU VFXTH
Fixed Destination Number: 0265 2630555
Enable this flag if you want to

block the IC calls received
without CLI on this SIP Trunk
g
defined here
(with CLI)
(without CLI)
Define fixed
d ti ti number
destination
b
on which you want
to route the call
Define destination

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
SIP
FXO
0265 2630555
SETU VFXTH
CallingNumber
Destination Number
02652630555
471
02226471110
472
Route on the basis of Calling Party Number

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

0265 2630555
SIP
FXO
02652630555@
203.88.142.221
0265 2630555
SETU VFXTH
203.88.142.221
defined here
1 Route all IC
1.
calls (with CLI)
2. Route all IC
calls (without
CLI)

Incoming calls as to the
called party number
Define destination
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
203.88.142.218
ReceivedCalledParty
3001
ReceivedCallingParty
2001
2002
2003
2001
3001
IP
Network
3001
3002
3003
SETUVFXTH
SETU
VFXTH
INDIA
115.118.161.164
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

calls,
ll enable
bl flag
fl to
t
this trunk
Call Block on SIP Trunk
For OG Call we can

calls,
ll enable
bl flag
fl to
t
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:192.168.50.161:5060
Source: 115.118.161.163:5060
Source:115.118.161.163:5060
STUNResponse
To:115.118.161.163:5060
Payload:115.118.161.163:5060
STUNServer
STUNResponse
To:192.168.50.161:5060
Payload:115.118.161.163:5060
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
Program Router Public

IP Address here
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
192.168.200.210
WAN
203.88.142.221
SETU VFXTH
IP: 192.168.200.195
G/W : 192.168.200.210
Private IP
203.88.142.218
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)
Apply ANT on FXO Port and

program the number list for dialed
aandd substitute
subst tute number
u be string
st g
Apply ANT on SIP Trunk and

program the number list for dialed
and substitute number string
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
Apply black listed caller

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

Set filter parameters
for CDR here
Click here to clear

all call records
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
CDR can also be

viewed from JEEVES
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
203.88.142.221
Ethernet
115.118.161.163
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
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
Enable apply flag in

SIP trunk to use
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
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 abc.com
b
andd destination
d i i number
b for
f call
ll
forward is *1239974 then it will be replaced by
9974@abc.com
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

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
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
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
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

call disconnection using
access code

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 192.167.100.1 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
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

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
config.zip having configuration
files & click on Restore
Click on save to get the

configuration backup
Browse the ZIP file with file name

firmware.zip 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 |
Phone : +91 265 2630555
Fax : +91 265 2636598
Training Querries: training@matrixcomsec.com
Training Contact : +91 9724341602
Technical Support Hours IST:
America | 19:00 to 03:00 | skype: support_matrixamerica | Ph. : +91 7600014634 / 9898074305 |
Europe | 13:00 to 21:00 | skype: support_matrixeurope | Ph. : +91 7600014834 |
AsiaAfricaMiddleEast | 09:00 to 18:00 | skype: support_matrixaame| Ph. : +91 9974044583 |
Domestic Night Support | 14:00 to 22:00 | skype : support_matrixindia | Ph. : +91 9724341600 |
Domestic Day Support | 09:00 to 06:00 | Ph. : +91 8128983042 / 8128983142 |
Demo | 09:00 to 06:00 | skype : demo_matrix| Ph. : +91 8128690107

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HEARTY

Installation Dos & Don'ts

Programming Using Phone

Incoming Call Management

Outgoing Call Management

A Versatile VoIP-FXO-FXS Gateway

A Gateway that provides voice service over IP network

An effective and flexible solution for accessing Internet

Developed to fulfill requirements of SOHO (Small Office

Total 32 SIP Trunks supported in all Configurations

SETU VFXTH1616 Hardware

VoIP module : CODEC IC &

400 ms RED on - Incoming Ring Event

Dust Proof, Moisture Free Location

Away from electromagnetic Sources

Path to Static Charges

Stable Mains Supply

Proper Mains Earth

Proper Telecom Earth

Stand Alone Application

In Front of PBX Application

Behind the PBX Application

Analog Extension PBX Over IP Application

Peer to Peer Callingg

PSTN Call over IP (Long Distance

PROGRAMMING USING PHONE

Programming Using Phone

Certain parameters of SETU VFXTH can be

You can configure

11 IP Address #* (To change IP Address)

12 Subnet Mask #* (To change Subnet Mask)

10 Code #* (to change the connection type)

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

51 Reverse SE Password #* (To Restore

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

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

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

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

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

SETU VFXTH : Configuration

SETU VFXTH : Configuration

SETU VFXTH: Configuration

Click on Start Internet

Enter Ethernet Port IP

Enter SE Password for

Network Port Parameters

This parameters can be programmed as per existing data network

Network Port Parameters

INCOMING CALL MANAGEMENT

Incoming Call Route

Destination Number Determination on FXO Port

Incoming Call Route

Destination Number Determination on FXO Port

Without Any Destination Number

Without Any Destination Number

Without Anyy Destination Number

Select route for all

Route To a Fixed Destination Number

Route To a Fixed Destination Number

Fixed Destination Number: 471

Route To a Fixed Destination Number

Enable this flag if you

Route on the basis of Callingg Partyy Number