matrix voip user manual

7/21/2019 matrix voip user manual

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HEART ELCOME !



“VOIP-FXO-FXS GATEWAY”



Agenda

• Overview

• Interfaces

• Port Configuration

• LED Indications

• Installation Do’s & Don'ts

• Applications



Agenda

• Programming Using Phone

• Programming Using PC

• Outgoing Call Management

• Advance Settings

• eatures

• Maintenance

• Status



O ER IE



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



INTERFACES



Interfaces

USB(Future Use)

Router

Ada tor

PC 24V DC

@ 2.5A60 W



CONFIGURATIONS



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



HARD ARE ARCHITECTURE



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





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



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



ys em

LED (STS)



DO’S &DON’TS



Installation Do’s

• Dust Proof, Moisture Free Location

• Away from electromagnetic Sources

• Ventilated Location

•

• Stable Mains Supply

• Proper Mains Earth

• Proper Telecom Earth



’



APPLICATIONS



Stand Alone A lication

2001

FXS1…FXS16

FXO1…FXO162016

PSTN

NetworkEthernet

Stand

Alone

Call

Possibilities

1. FXS IP Network

BroadbandModem/Router

Network



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


IP

Network

Networks of VFXTH



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


IP

Network

Trunk

of

VFXTH



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



Peer to Peer CallinIP

SETU

VFXT1616

SETU

VFXT1616

2001FXS1…

FXS16

FXS1…

FXS16

FXO1…

FXO16

FXO1…

FXO16

Ethernet Ethernet



PSTN Call over IP (Long Distance

2001 IP

SETU

VFXT1616SETU

VFXT1616

2001

FXS1…

FXS16

FXS1…

FXS162016

FXO1…

FXO16

FXO1…

FXO16

Ethernet Ethernet

Mumbai Delhi



PROGRAMMING USING PHONE



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



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



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)



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)





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



SETU VFXTH : Configuration

Web Jeeves Login from Public Network

n erne 203.88.123.231

SETU VFXTH is located

PC with internet

connection



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



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



Click on Start Internet

Explorer (Any Browser)



Programming

Enter Ethernet Port IPAddress of SETU VFXTH



Login Page

Enter SE Password forLogin (Default: 1234)



Home Page



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



Network Port Parameters

Select connection type of SETU

VFXTH and accordin to the

connection type program the IP details



INCOMING CALL MANAGEMENT - SIP TRUNK

-



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



DETERMINATION



FXO PORT



Destination Number Determination on FXO Port

Incoming Call Route



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



• 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



FXOFXS

022 2631725

SETU VFXTH



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)



Route To a Fixed Destination Number


• 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




FXO

SIP

471@matrix- 0265 2630555p x. yna as.org

SETU VFXTH





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



Route on the basis of Callin Part Number


• 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




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



Route on the basis of Calling Party Number


Incoming calls as on

the basis of calling party number


defined here

1. Route all IC

ca s w

CLI)2. Route all IC

calls (without




Pro ram the callin

number based table with

calling party number

and destination number



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



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



After Answering the call & collecting the digits

Select route for all Incoming

calls as after answering the

call and collecting the digits


defined hereDefine destination.

calls (with CLI)2. Route all IC

calls (without

port for routing calls



SIP TRUNK

Destination Number Determination on SIP Trunk




Incoming Call Route

options 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



• 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



SIPFXS

022 2631725

SETU VFXTH






Incomin calls as Without


defined here

1. Route all ICany Destination numbercalls (with

CLI)

2. Route all IC

calls without

CLI)

Define destination







• 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





SIPFXO

0265 2630555 471@matrix-

pbx.dynalias.org

SETU VFXTH





Enable this flag if you want to

2 different routin s

block the IC calls received

without CLI on this SIP Trunk

defined here


(with CLI)


Define fixed

(without CLI)

es na on num er

on which you want

to route the callDefine destination







• 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





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



Incoming calls as on

the basis of calling

party number


defined here

1. Route all IC

2. Route all IC

calls (without

CLI)

Define destination





C

number based table withcalling party number

and destination number

To the Called Part Number



• 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




0265 2630555

SIPFXO

0265 2630555

02652630555@

203.88.142.221

SETU VFXTH203.88.142.221





called party number 2 different routings

defined here

.

calls (with CLI)

2. Route all IC

calls (without

Define destination




DETERMINATION

Destination Port Determination



•

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



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 ort determination

options on SIP Trunk



DETERMINATION – FIXED

Fixed



Program the routing groupClick on Edit to

calls on the SIP trunkof routing group

Fixed



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



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



DESTINATION PORTDETERMINATION – ON THE

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 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



Enter the destination

group is to be programmed

Program the routing group

and fallback routing group

for the destination number

defined above



DESTINATION PORTDETERMINATION – ON THE BASIS

On the Basis Of Callin Part Number



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



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



OUTGOING CALL MANAGEMENT-

-- SIP TRUNK

Call Block on FXS Port



For OG Call we can

allow or block outgoingcalls, enable flag to

this trunk



Call Block on SIP Trunk



For OG Call we canallow or block outgoing

ca s, ena e ag o

Block the Outgoing from

this trunk



STUN

STUN (Simple Traversal of UDPs

through NATs)



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



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



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



Select NAT type as STUN if you want

to use a ress etc e us ng

STUN



Status page will display the IP

a ress, por num er antype fetched using STUN



ROUTER PUBLIC IP ADDRESS

Router’s Public IP Address



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



Public IP address




Program Router Public

IP Address here




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



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



Router’s

Network

Parameters

*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.

Router Configuration: Example



Port Forwarding:

Router’s SIP an RTP Port

forwarded to Private IP ofSETU VFXTH

*Linksys

is

a

wholly

owned

subsidiary

of

Cisco

Systems,

Inc.



Access Codes

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



• 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



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



• 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



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

ort & ro ram 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

Automatic Number Translation

• This feature is used to translate the dialed number strin to



• 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)




Apply ANT on FXO Port and

program the number list for dialed

and substitute number strin




Apply ANT on SIP Trunk and

program the number list for dialed

and substitute number string




Pro ram dialed and substitute number

strings in the number list table

Black Listed Callers

• ‘ ’



•

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




Apply black listed caller

feature on selected SIPtrunk and define the

number list for the same




Pro ram the number list with

the CLI of black listed callers

Call Detail Record (CDR)

• ’



,

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

•


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



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


• 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

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


Set filter parameters



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




CDR can also be

PIN Authentication

• PIN authentication is a security feature to restrict access to the



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



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



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



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



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



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



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



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




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




Enable apply flag in

SIP trunk to usedigest authentication




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



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



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



,

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




• 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

[email protected]




Define prefix and domain

name in the table



Disconnect Tone



Enable disconnection

tone detection here

Disconnect Tone



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



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



Click here to Edit

entry of the table

entry to the table Click here to delete

entry from the table



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 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



Enable required

of service on

particular FXS port

Supplementary Services



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



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



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



Select the

subscriber typeof your choice

FXS Port

Si nalin Loo Start

Connector RJ-45



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



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



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

• It is programmable from 01 to 99 seconds (Default: 15

seconds)

FXS

•



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



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



FXO Port



Hardware settings

on FXO port

FXO Port



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



• 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


•

In speech with the current call



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




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



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




Enable the flag to allowca sconnec on us ng

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. . 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



SIP trunk or SIP

trunk group can be

defined IP dialing

SIP Timers

•

•

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



• 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)


• 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



• 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


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



• 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


SIP Over TCP

•

send/receive the SIP messages over TCP



• 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



values according

to the requirement



Firmware/Configuration

Browse the ZIP file with file name‘config.zip’ having configuration

es c c on es ore

configuration backup



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

,



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



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



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



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



• 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



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



c o ac ory

default the gateway

Soft Restart



Click OK to Restart

SETU VFXTH



Network



Network status

with IP details

FXO Port

FXO Port status whether

Li i t d t



Line is connected or not

SIP Trunk



SIP trunk statusdisabled, active etc.

eason o a ure n

case Registration failed

Firmware



Firmware Version

MATRIX COMSEC PVT. LTD.



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