Download - matrix voip user manual
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HEART ELCOME !
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“VOIP-FXO-FXS GATEWAY”
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Agenda
• Overview
• Interfaces
• Port Configuration
• LED Indications
• Installation Do’s & Don'ts
• Applications
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Agenda
• Programming Using Phone
• Programming Using PC
• Outgoing Call Management
• Advance Settings
• eatures
• Maintenance
• Status
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O ER IE
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Overview
• A Versatile VoIP-FXO-FXS Gateway
• A Gateway that provides voice service over IP network
• An effective and flexible solution for accessing Internet
based telephone services & corporate Internet systems
• Developed to fulfill requirements of SOHO (Small Office
Home Office) users & small/ medium scale enterprises
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INTERFACES
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Interfaces
USB(Future Use)
Router
Ada tor
PC 24V DC
@ 2.5A60 W
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CONFIGURATIONS
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VFXTH Configurations
Configurations VoIP
Channels
FXO
Ports
FXS
Ports
FXO Ports
Label
FXS Ports
Label
VFXTH0016 16 0 16 0 P01-P16
VFXTH0024 24 0 24 0 P01-P24
-
VFXTH0800 8 8 0 P01-P08 0
-
VFXTH2400 24 24 0 P01-P24 0
VFXTH3200 32 32 0 P01-P32 0
VFXTH0808 16 8 8 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
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HARD ARE ARCHITECTURE
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SETU VFXTH1616 Hardware
SLT Modules:
Each Module
supports 2TWT
ModulesVoIP module : CODEC IC &
SDRAM. Total 4 such Modules 8 BIT
be connected(Total 8)
(Total ) Each supporting 8 channels RAM
Power
Supply
OP V :
PROCESSOR
FLASH
RAM
Input
+3.3V,
-27V
-87V
CPLD
Power Jack
24 V, 2.5A
Ethernet Port
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LED Indications
Total 34 LEDs in SETU VFXTH1616
(Continuous Green)
32 Port LEDs : 16 (P01-P16) FXO Port LEDS
16 (P17-P32) FXS Port LEDS
At Initialization:
P01-P32 : OFF
A ter approx 16 sec P01-P03 G ow Cont nuous ReAfter approx 20 sec remaining P04-P32 Glow Red Continuous
After 5 Sec : P01 - P32 LED will be Off
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LED Indications
32 FXO/FXS Ports LED indications during normal functioning
ont nuous y ort e sa e
-
200 ms off -
400 ms RED on -
ms o
( 2 Blinks)
400 ms on- Off-Hook Event Dialin State
400 ms off
(continuous)
ont nuous n
Red
peec
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ys em
LED (STS)
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DO’S &DON’TS
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Installation Do’s
• Dust Proof, Moisture Free Location
• Away from electromagnetic Sources
• Ventilated Location
•
• Stable Mains Supply
• Proper Mains Earth
• Proper Telecom Earth
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’
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APPLICATIONS
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Stand Alone A lication
2001
FXS1…FXS16
FXO1…FXO162016
PSTN
NetworkEthernet
Stand
Alone
Call
Possibilities
1. FXS IP Network
BroadbandModem/Router
Network
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In Front of PBX A licationPBX
SETU
VFXT1616
FXS1…FXS16
FXO1…FXO16
FXS1…FXSN
PSTN
NetworkEthernet
FXS
ports
of
VFXTH
are
connected
to
FXO
Ports
of PBX.
Extensions
of
PBX
thus
can
avail
the
PSTN
&
VoIP
BroadbandModem/Router
IP
Network
Networks of VFXTH
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Behind the PBX A lication
2001
SETU
VFXT1616
PBXFXS1…FXS16
FXO1…FXO162016
PSTN
N/W
EthernetFXO
ports
of
VFXTH
are
connected
to
FXS
FXS1…FXSN
Ports
of
PBX.
Extensions
of
VFXTH
BroadbandModem/Router
IP
Network
Trunk
of
VFXTH
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Analog Extension PBX Over IP Application
SETU
VFXT1616SETU
VFXT1616
2016FXO1…
FXO16FXS1…
FXS16 Broadband
PSTN
N/W
FXO1…
FXO16FXO1…
FXO16
Modem/Router
Ethernet PBX FXS1…FXSN
E t h e r n e t
IP
Network BroadbandModem/Router
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Peer to Peer CallinIP
SETU
VFXT1616
SETU
VFXT1616
2001FXS1…
FXS16
FXS1…
FXS16
FXO1…
FXO16
FXO1…
FXO16
Ethernet Ethernet
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PSTN Call over IP (Long Distance
2001 IP
SETU
VFXT1616SETU
VFXT1616
2001
FXS1…
FXS16
FXS1…
FXS162016
FXO1…
FXO16
FXO1…
FXO16
Ethernet Ethernet
Mumbai Delhi
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PROGRAMMING USING PHONE
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Programming Using Phone
• Certain parameters of SETU VFXTH can be
configured by dialing system commands from a
e ep one connec e o e por
• You can configure certain network parameters like
IP address, Subnet Mask, Connection Type, set the
system to e au t an a so v ew current a ress,
Subnet Mask, Connection Type, DNS and Gateway
address by dialing system commands
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SE Login
• Connect Analog Phone to FXS port of SETU VFXTH
• OFF - Hook the phone
• Hear Dial Tone [Toooooooooooooooooooooo]
“ ” –
• Default SE Password is “1234”
• Enter System Commands to perform different functions
• Dial “00#*” to Exit from Programming mode
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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 Ta
• 51 – Reverse SE Password – #* (To Restore
Factory defaults)
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Commands
• 21 – #* (To view IP Address) & go 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
• – * –
• 27 – SIP Trunk Number (1 – 9) – #* & go On – hook
(To view the status of SIP Trunk)
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SETU VFXTH : Configuration
Web Jeeves Login from Local Network
Network Switch
192.168.50.200
SETU
VFXTH
is
located
on Local IP
192.168.50.33
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SETU VFXTH : Configuration
Web Jeeves Login from Public Network
n erne 203.88.123.231
SETU VFXTH is located
PC with internet
connection
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SETU VFXTH: Configuration
Web Jeeves Login from Public Network LAN:
192.168.1.1
WAN: IP
:
192.168.1.151. . .Subnet : 255.255.255.0
Gateway : 192.168.1.1
Router s port:80 is
forwarded
to
IP
Address
of
SETU VFXTH
PC with internet
connection
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Programming
• u – n e server
•
•
Accessible using any web browser
• Default IP of Ethernet Port is 192.168.001.147
• Default SE password is 1234
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Click on Start Internet
Explorer (Any Browser)
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Programming
Enter Ethernet Port IPAddress of SETU VFXTH
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Login Page
Enter SE Password forLogin (Default: 1234)
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Home Page
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Network Port Parameters
• This parameters can be programmed as per existing data network
• Connection type :
. tat c: a ress, u net mas ateway ress ass gne
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
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Network Port Parameters
Select connection type of SETU
VFXTH and accordin to the
connection type program the IP details
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INCOMING CALL MANAGEMENT - SIP TRUNK
-
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Incoming Call Route
• ,
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
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DETERMINATION
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FXO PORT
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Destination Number Determination on FXO Port
Incoming Call Route
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Destination Number Determination on FXO Port
• W t out any Dest nat on Num er
• On the basis of Callin Part Number
• After answering the call and collecting the digits
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• Incomin call on the FXO ort
• All calls received on the FXO port are directly
routed to the fixed destination port, configured for
this ort, re ardless of the destination number
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FXOFXS
022 2631725
SETU VFXTH
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Without An Destination Number
Select route for all
Incoming calls as Without
2 different routings
defined here
1. Route all IC any est nat on num ercalls (with
CLI)
2. Route all IC
calls without
CLI)
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Route To a Fixed Destination Number
• Incomin call on the FXO ort
• Call is routed to the Fixed destination number
programmed on that particular trunk line using the
Destination ort ro rammed for that trunk
• Destination port can be FXS port, FXO port or SIP
Trunk
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Route To a Fixed Destination Number
FXO
SIP
471@matrix- 0265 2630555p x. yna as.org
SETU VFXTH
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Route To a Fixed Destination Number
2 different routings
defined here
Enable this flag if you
want to block the IC
calls received without
Define destination
port for routing calls.(with CLI)
2. Route all IC calls
(without CLI)
CLI on this FXO port
Define fixed
es na on num er
on which you want
to route the call
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Route on the basis of Callin Part Number
• Incomin call on the FXO ort
• Calls are routed to a specific number according to
the calling party number
• ,
SETU VFXTH will match the calling party number
with the entries of the calling party number basedtable, if a match is found, the call is routed to the
destination number
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Route on the basis of Callin Part Number
FXOSIP
471@matrix- 0265 2630555pbx.dynalias.org
Calling
Number Destination Number
SETU VFXTH
02652630555 471
02226471110 472
R t th b i f C lli P t N b
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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
ca s w
CLI)2. Route all IC
calls (without
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Route on the basis of Callin Part Number
Pro ram the callin
number based table with
calling party number
and destination number
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After Answerin the call & collectin the di its
• ncom ng ca on e por
• Incomin calls are answered and dial tone is la ed
to the caller, allowing the caller to dial the desired
num er
• The number dialed b the caller is considered as the
destination number and dial it out using the
es na on por programme
Af A i h ll & ll i h di i
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After Answerin the call & collectin the di its
471
FXOSIP
471@matrix-
0265 2630555
SETU VFXTH
pbx.dynalias.org
After Answering the call & collecting the digits
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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 hereDefine destination.
calls (with CLI)2. Route all IC
calls (without
port for routing calls
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SIP TRUNK
Destination Number Determination on SIP Trunk
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Destination Number Determination on SIP Trunk
Incoming Call Route
options on SIP trunk
Destination Number Determination on SIP Trunk
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Destination Number Determination on SIP Trunk
• W t out any Dest nat on Num er
•
On the basis of Callin Part Number
• After answering the call and collecting the digits
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• Incomin call on the SIP Trunk
• All calls received on the SIP Trunk are directly
routed to the fixed destination port, configured for
this ort, re ardless of the destination number
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SIPFXS
022 2631725
SETU VFXTH
Without An Destination Number
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Without An Destination Number
Select route for all
Incomin calls as Without
2 different routings
defined here
1. Route all ICany Destination numbercalls (with
CLI)
2. Route all IC
calls without
CLI)
Define destination
port for routing calls
Route To a Fixed Destination Number
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Route To a Fixed Destination Number
• Incomin call on the SIP Trunk
• Calls are routed to the Fixed destination number
programmed on that SIP trunk using the Destination
ort ro rammed for that SIP trunk
• Destination port can be FXS port, FXO port or SIP
Trunk
Route To a Fixed Destination Number
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Route To a Fixed Destination Number
SIPFXO
0265 2630555 471@matrix-
pbx.dynalias.org
SETU VFXTH
Route To a Fixed Destination Number
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Route To a Fixed Destination Number
Enable this flag if you want to
2 different routin s
block the IC calls received
without CLI on this SIP Trunk
defined here
1. Route all IC calls
(with CLI)
2. Route all IC calls
Define fixed
(without CLI)
es na on num er
on which you want
to route the callDefine destination
port for routing calls
Route on the basis of Callin Part Number
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Route on the basis of Callin Part Number
• Incomin call on the SIP Trunk
• Calls are routed to a specific number according to
the calling party number
• ,
SETU VFXTH will match the calling party number
with the entries of the calling party number basedtable, if a match is found, the call is routed to the
destination number
Route on the basis of Callin Part Number
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Route on the basis of Callin Part Number
SIPFXO
471@matrix-0265 2630555 pbx.dynalias.org
Calling
Number Destination Number
SETU VFXTH
02652630555 471
02226471110 472
Route on the basis of Calling Party Number
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Incoming calls as on
the basis of calling
party number
2 different routings
defined here
1. Route all IC
2. Route all IC
calls (without
CLI)
Define destination
port for routing calls
Route on the basis of Callin Part Number
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C
number based table withcalling party number
and destination number
To the Called Part Number
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• ncom ng ca on e run
• Incomin calls are routed to a desired number
depending upon the called number received in the
o reques o e message
To the Called Part Number
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0265 2630555
SIPFXO
0265 2630555
02652630555@
203.88.142.221
SETU VFXTH203.88.142.221
To the Called Part Number
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Select route for all
called party number 2 different routings
defined here
.
calls (with CLI)
2. Route all IC
calls (without
Define destination
port for routing calls
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DETERMINATION
Destination Port Determination
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•
determining the destination port for the calls
originated on FXS Port, FXO Port and SIP trunks,
they are:
1. Fixed
.
3. On the basis of calling party number ( Not
Supported on FXS Port)
Destination Port Determination on FXS Port
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Destination port determination
options on FXS port
Destination Port Determination on FXO Port
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Destination port determination
options on FXO port
Destination Port Determination on SIP Trunk
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Destination ort determination
options on SIP Trunk
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DETERMINATION – FIXED
Fixed
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Program the routing groupClick on Edit to
calls on the SIP trunkof routing group
Fixed
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rogram
number to be sent on
destination port
Program Group
Member for
routing group
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
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2001 203.88.142.2183001 Received Called Party
3001
2002
2003
Received Calling Party
2001
3001IP
Network
20013001
3002
3003
INDIA115.118.161.164
SETU VFXTH
USA
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DESTINATION PORTDETERMINATION – ON THE
On the Basis Of Destination Number
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Program the Destination
number and routing group
in the destination number
based routing table
Click on Add to add Click on Delete to
t e new entry or
destination number
e e e e se ec e en ry
for destination number
On the Basis Of Destination Number
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Enter the destination
group is to be programmed
Program the routing group
and fallback routing group
for the destination number
defined above
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DESTINATION PORTDETERMINATION – ON THE BASIS
On the Basis Of Callin Part Number
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Program the calling
in the calling number
based routing table
Click on Add to add
the new entr for
Click on Delete to
delete the selected entr
Calling Number for Calling Number
On the Basis Of Callin Part Number
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Enter the Calling number
or w ic t e routing group
is to be programmed
Program the routing
group and fallback
routing group for the
Calling number
defined above
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OUTGOING CALL MANAGEMENT-
-- SIP TRUNK
Call Block on FXS Port
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For OG Call we can
allow or block outgoingcalls, enable flag to
this trunk
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Call Block on SIP Trunk
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For OG Call we canallow or block outgoing
ca s, ena e ag o
Block the Outgoing from
this trunk
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STUN
STUN (Simple Traversal of UDPs
through NATs)
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through NATs)
• When the VoIP port (WAN) is located behind a
the Public Internet
• STUN specifies the mechanism required for NAT
raversa n messages. server ac a es
traversing through most NATs except symmetric
NATs
Illustration of STUN
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STUN Request
STUN Request
Source:192.168.50.161:5060
STUN Response
STUN Response
. . .
To:115.118.161.163:5060Payload:115.118.161.163:5060 To: 192.168.50.161:5060
Payload:115.118.161.163:5060
STUN
Server
STUN
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Program the STUN Server Address; Listening Port of
STUN Server (1024-65535) Default port : 03478; Enable
e ag se or e c e us ng por
required to be fetched by STUN else disable when PortForwarding in the Router is done for SIP messages
STUN
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Select NAT type as STUN if you want
to use a ress etc e us ng
STUN
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Status page will display the IP
a ress, por num er antype fetched using STUN
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ROUTER PUBLIC IP ADDRESS
Router’s Public IP Address
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Port Forwarding :
Since STUN doesn’t work with symmetric NAT , as an
a erna ve o or orwar ng can e one n e
router and Router’s Public address that is configured
can be used as Source Port IP Address
Router Public IP Address
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Public IP address
Router Public IP Address
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Program Router Public
IP Address here
Router Public IP Address
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Status page will display
the Router Public IP
address programmed in
e sys em parame er page
P2P Call One Device is on Public IP and
t er ev ce nsta e e n
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Router separates
Private and Public Network
Port Forward in
Router
203.88.142.218
192.168.200.210
Public IPLAN WAN
203.88.142.221
SETU VFXTH
IP: 192.168.200.195
G/W : 192.168.200.210
Private IP
Router Configuration: Example
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Router’s
Network
Parameters
*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router Configuration: Example
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Port Forwarding:
Router’s SIP an RTP Port
forwarded to Private IP ofSETU VFXTH
*Linksys
is
a
wholly
owned
subsidiary
of
Cisco
Systems,
Inc.
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Access Codes
• Access code is a string of digits dialed to use a feature
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• Access code is a string of digits dialed to use a feature
• SETU VFXTH users can access the features and facilities by
dialin the access code assi ned to them from a hone.
SE/User can
.
2. Access Supplementary feature
3. Enter into the programming mode
,
you can change it to suit your preferences
Access Codes
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Access codes can be
changed from here
Access Codes
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Access codes can be
changed from here
Allowed – Denied Numbers
• This feature provides the flexibility to allow or deny dialing of
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• This feature provides the flexibility to allow or deny dialing of
a particular number or a set of numbers from a particular port
or all orts
• 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
number list – 8)
Allowed – Denied Logic on FXS Port
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Apply allowed denied list on FXS
Port & program the number list
for allowed & denied numbers
Allowed – Denied Logic on FXO Port
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Apply allowed denied list on FXO
ort & ro ram the number list for
allowed & denied numbers
Allowed – Denied Logic on SIP Trunk
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Apply allowed denied list on SIP
trunk & program the number list for
Automatic Number Translation
• This feature is used to translate the dialed number strin to
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• This feature is used to translate the dialed number strin 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
• ANT feature is applied on destination ports (On all SIP
trunks and FXO Ports)
Automatic Number Translation
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Apply ANT on FXO Port and
program the number list for dialed
and substitute number strin
Automatic Number Translation
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Apply ANT on SIP Trunk and
program the number list for dialed
and substitute number string
Automatic Number Translation
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Pro ram dialed and substitute number
strings in the number list table
Black Listed Callers
• ‘ ’
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•
enables you to block incoming calls from specific numbers
an a resses on e run s
• This feature is applicable on source port only
• To use this feature, user must configure the numbers of
unwan e ca ers n a num er s
• Enable the Reject Calls from Blacklisted Caller check box
on the SIP trunks on which you want to apply this feature
Black Listed Callers
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Apply black listed caller
feature on selected SIPtrunk and define the
number list for the same
Black Listed Callers
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Pro ram the number list with
the CLI of black listed callers
Call Detail Record (CDR)
• ’
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,
gateway’s usage when call was made
• Maximum of 2000 call record entries can be stored
• User can set different filters as required and generate Call
Detail Record (CDR) report
•
Call Detail Record (CDR)
• It is possible to get following details of a call with CDR
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1. Date of call origination
2. Time of call origination. a ng num er
4. Called number
5. Duration of call
6. Source port
7. Destination port
8. Disconnected by
9. Cause
. num er
11. Remarks
Call Detail Record (CDR)
• Below mentioned filter can be programmed for CDR
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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
4. Calls made at a particular time
5. Calls of a certain duration
7. Calls of certain calling party numbers.
9. Calls made without PIN authentication
Call Detail Record (CDR)
Set filter parameters
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Click here to clear
all call records
Call Detail Record (CDR)
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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)
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CDR can also be
Call Detail Record (CDR)
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CDR can also be
PIN Authentication
• PIN authentication is a security feature to restrict access to the
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system and prevent possible misuse of resources
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
th i di th ti ti d
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their corresponding authentication passwords
• If PIN authentication is enabled on source ort SETU VFXTH
answers the Incoming call and plays a feature tone, it waits for the
caller to dial the PIN number and assword, 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
make two more attem ts if the caller fails to dial correct PIN and
password in all attempts, the system disconnects the call
PIN Authentication – FXO Port
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Select routing type
‘a ter answer ng t ecall and collecting
the digits’ for PIN
authentication
feature to use
na e s ag
for prompting
caller to enter PIN
PIN Authentication
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Enter PIN number & PIN password,
system checks PIN entered by the callerdurin call with the entries in the PIN
authentication table, if match found then
only the call will be processed further
Peer to Peer Dialing
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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
very attractive mode of inter – branch or intra – office voice
communication
Peer to Peer Callin
IPSETU
VFXTH SETU VFXTH
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3001
2001
IP
NetworkLocation
– A Location
– B
FXS1…
FXS16
FXS1…
FXS163XXX
FXO1…FXO16
FXO1…FXO16
Ethernet Ethernet
203.88.142.221 115.118.161.163
Peer to Peer Dialing
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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 here to delete
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
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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
Password – in the INVITE message are authenticated and accepted
by the recipient, the originator and recipient are connected
• You ma use the di est authentication to restrict access to SETU
VFXTH to specific callers, prevent unwanted or malicious calls
Digest Authentication
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1. SETU VFXTH will challenge the identity of the calling party
. en e ca ng par y sen s s cre en a s,
authenticates the credentials by matching it with its Digest
ut ent cat on ta e3. 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
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Enable apply flag in
SIP trunk to usedigest authentication
Digest Authentication
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Enter Digest credentials (User ID
an ser asswor o ca ng par y
Static Routing
• Static Routin Table is re uired when ou have more than one
t (G t ) i t k d t SETU
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router (Gateway) in your network and you want SETU
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
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Program the static routing table withthe 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
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,
applicable only when the destination port is SIP
• SETU VFXTH supports multiple SIP trunks & FXS ports,
,
routes the call to the IP network using the SIP trunk
.
only numbers not domain names, therefore it becomes
which the FXS user can dial
Prefix to Domain Name Conversion
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• This table is not checked for making an outgoing call, but it
is checked when some FXS port has set call forward and
onl number is ro rammed or user is doin blind transfer
• For example prefix in the table is programmed as *123 and
oma n name as a c.com an est nat on num er or ca
forward is *1239974 then it will be replaced by
Prefix to Domain Name Conversion
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Define prefix and domain
name in the table
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Disconnect Tone
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Enable disconnection
tone detection here
Disconnect Tone
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Program the disconnecttone cadence here
Emergency Numbers
• SETU VFXTH su orts dialin of emer enc numbers from
all ports, Emergency numbers and their respective routing
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po s, e ge cy u be s d e espec ve ou g
• User can configure up to 10 numbers of emergency services
such as ambulance, fire brigade, police etc.
• , ,
in the emergency number table
Emergency Numbers
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Click here to Edit
entry of the table
entry to the table Click here to delete
entry from the table
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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
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at SIP services provider in case of SIP account calling
• SETU VFXTH offers following telephony features, which they can
access b dialin 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
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Enable required
of service on
particular FXS port
Supplementary Services
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Enable the
services after
enabling thefeature in COS
Subscriber Type
• en s n er ace w serv ce prov er
server – ITSP or other PBX that supports supplementary
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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
ort connected to the s stem
Subscriber Type
•
supported by the other PBX, you can access the service
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prov er ea ures y a ng , you w no e a e o
access the local features of SETU VFXTH
• Select Gateway if you want to use supplementary services
su orted b the SETU VFXTH in the atewa mode ou
will also be able to access the supplementary services of the
Subscriber Type
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Select the
subscriber typeof your choice
FXS Port
Si nalin Loo Start
Connector RJ-45
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Off-Hook Line Impedance 600 Ω / 900 Ω / Complex No. of Long Loop Extension 4
Loop Limit 1800 (Max) Excluding Telephone
SetOn-Hook Voltage (Tip/Ring) -48 V
Off-Hook Current 25 mA (Max)
Ringing Voltage Trapezoidal 60 VRMS/25Hz and
FXS Port
REN 3
e ec on - .
CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore
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ro ec on ver o age econ ary ro ec on
Return Loss >18 dB
Transmission Level
Ad ust
Tx Gain : -3dB to +6dB ;
Rx Gain : -3dB to +6dB
Answer Signaling on
FXS
Battery Reversal
Disconnect Signaling
on FXS
Battery Reversal & Open Loop
Disconnect
FXS Port
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Hardware settings
on FXS port
FXS Port
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General settings
on FXS port
FXS Port
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General settings
on FXS port
FXS Port
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First Digit & Inter
Digit wait timer
FXS
First Digit Wait Timer:
• Signifies the time for which the system waits for
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receiving a first digit after going off-hook from FXS
port
• On expiry of this timer, system will give error tone to
• It is programmable from 01 to 99 seconds (Default: 15
seconds)
FXS
•
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Signifies the time period between 2 consecutive digitswhile the system is receiving the digits from caller
•On expiry of this timer, ATA1S will process the digits
dialed so far b the user
•it is programmable from 01 to 99 seconds (Default: 5
seconds)
FXO Port
Longitudinal Balance >50 dB
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ransm ss on eve ust x a n: - to +
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
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FXO Port
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Hardware settings
on FXO port
FXO Port
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on FXO port
Making a new call using access code
• This feature enables callers to disconnect the current call and make a
new ca us ng w t out gett ng sconnecte rom
the system
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• 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 whenAfter answering the call and collecting digits is selected as the
est nat on num er eterm nat on met o
• If you have enabled Connect source port when number is outdialed on the FXO port, you will not be able to provide this feature
to callers
Making a new call using access code
•
In speech with the current call
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a
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 s eech dial #91 a ain to make another new call
Making a new call using access code
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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
releases the ort en a ed in the call
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releases the ort en a ed in the call
• This feature is applicable only when destination number
collecting digits
•
dialed on the FXO port or have enabled Connect source port when
,
provide this feature to users
Disconnecting a call using access code
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Enable the flag to allowca sconnec on us ng
access code
Disconnecting a call using access code
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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
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• IP number can be dialed with dot ’.’ as entered by ‘*’ while dialing it
• For e. . to dial IP address 192.167.100.1 dial as 192*167*100*1 from
the Phone at FXS
,
the system does not check the destination port determination method
,
through the SIP trunk or SIP group you configured for IP dialing
IP Dialing
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SIP trunk or SIP
trunk group can be
defined IP dialing
SIP Timers
•
•
SIP Provisional Timer
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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
Thi ti t t ft di INVITE t th
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• This time starts after sending INVITE message to the
called art and sto s on recei t of rovisional
response or final response or when the user goes ON-
oo , on exp ry o t e t mer t e ca s sconnecte
• The ran e of SIP INVITE Timer is 10 - 80 seconds
(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
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• 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
• The ran e of SIP Provisional Timer is 10 - 180 seconds
(Default: 60 Seconds)
General Re uest Timer
• It is the time for which VFXTH waits for the res onse
of a transaction request
Thi ti t t i iti ti t ti
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• This timer starts on initiating a transaction
• This timer stops on receipt of a response for the request
• On ex ir of timer the VFXTH clears the transaction
•The range of SIP Provisional Timer is 10 - 60 seconds
(Default: 20 Seconds)
SIP Over TCP
•
send/receive the SIP messages over TCP
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• SIP over TCP is applicable for both Proxy and Peer-
-
• By Default SIP messages transported over TCP
•
Disable the flag to send SIP messages over UDP
S stem Parameters
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values according
to the requirement
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Firmware/Configuration
Browse the ZIP file with file name‘config.zip’ having configuration
es c c on es ore
configuration backup
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configuration backup
Click on download to get
Browse the ZIP file with file name
‘firmware.zip’ having new
firmware files & click on Upgrade
Zip i e containing CDR
in .csv and .txt format
to upgra e t e system rmware
System Debug
• Debugs are logs of actions and events that take place on system,
these logs are useful for troubleshooting and system security
,
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client enables the system to send debug messages in Syslog
orma o e remo e ys og server on e ne wor
• Syslog uses the UDP as transport protocol
• To be able to use this feature, you must enable ‘Syslog’,
confi ure the S slo Server Address and define the server ort
on which the Syslog will listen for debug messages
System Debug
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Debu events
can be viewed
on the screen
System Debug
Program the IP address and
port number of PC/Laptop
h l i i ll d
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where Syslog server is installed
e ug or or : c ear e c ec
box to disable the debug for the port which is not needed
System Debug
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Click on Save Debu 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
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• A maximum 2MB of packets can be captured and stored in the system
• SETU VFXTH also supports filters and promiscuous mode for
• If promiscuous mode is enabled, SETU VFXTH will capture all
ne wor ra c an sa e en sys em w cap ure on y ra c
that is directly related to SETU VFXTH (to or from SETU VFXTH)
PCAP Trace
Click here to Enable
Enter the filter
details here
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Click here to Enable
Promiscuous mode
details here
Click here to start
the PCAP traceClick here to stop
the PCAP trace
captured save the trace
file on your PC/Laptop
Default System
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c o ac ory
default the gateway
Soft Restart
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Click OK to Restart
SETU VFXTH
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Network
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Network status
with IP details
FXO Port
FXO Port status whether
Li i t d t
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Line is connected or not
SIP Trunk
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SIP trunk statusdisabled, active etc.
eason o a ure n
case Registration failed
Firmware
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Firmware Version
MATRIX COMSEC PVT. LTD.
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|Vadodara | Gujarat | India |
Phone : +91 265 2630555
Fax : +91 265 2636598Training Querries: [email protected]
Training Contact : +91 9724341602
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