SETU VFXTH March 2014

Hearty Welcome!

Technical Training

SETU VFXTH

“VoIP – FXO – FXS Gateway”

Overview

Interfaces

Port Configuration

Hardware Architecture

LED Indications

Installation Do’s and Don'ts

Applications

Programming using Phone

Programming using PC

Incoming call Management

Outgoing Call Management

Advance Settings

Maintenance

Status

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

Overview

Interfaces

Overview

Port Configuration


LED Indications


Applications





Advance Settings

Maintenance

Status

Interfaces

Port Configuration

Overview

Interfaces


LED Indications


Applications





Advance Settings

Maintenance

Status

Configurations VoIP

Channels

FXO

Ports

FXS

Ports

FXO Ports

Label

FXS Ports

Label

SETU VFXTH0016 16 0 16 0 P01-P16

SETU VFXTH0024 24 0 24 0 P01-P24

SETU VFXTH0032 32 0 32 0 P01-P32

SETU VFXTH3200 32 32 0 P01-P32 0

SETU VFXTH0808 16 8 8 P01-P08 P09-P16

Total 32 SIP Trunks supported in all Configurations

SETU VFXTH Configurations


Overview

Interfaces

Port Configuration

LED Indications


Applications





Advance Settings

Maintenance

Status

SETU VFXTH1616 Hardware Architecture

Input Supply : DC Power Jack 24 V, 2.5A

Power Supply OP V : +5V +3.3V, -27V -87V

Ethernet Port

FXS Modules: Each Module supports 2 extensions to be connected (Total 8)

FXO Modules (Total 8)

VoIP module : CODEC IC & SDRAM. Total 4 such Modules

Each supporting 8 channels

32 BIT RISC PROCESSOR

FLASH 32MB

128 MB

RAM

CPLD

LED Indications

Overview

Interfaces

Port Configuration



Applications





Advance Settings

Maintenance

Status

Total 34 LEDs in SETU VFXTH1616

Power LED : At Power On Power LED will Turn On

(Continuous Green)

32 Port LEDs : FXO and FXS Port LEDS

At Initialization:

P01-P32 : OFF

After approx 16 sec P01-P03 Glow Continuous Red

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

Continuously Off Port Idle / Disable

400 ms Red on -

200 ms off -

400 ms RED on -

3000 ms off (2 Blinks)

Incoming Ring Event

400 ms on-

400 ms off (continuous) Red

Off-Hook Event (Dialing State)

Continuous On Red Speech

LED Indications

System

LED (STS)

Installations Do’s and Don’ts

Overview

Interfaces

Port Configuration


LED Indications

Applications





Advance Settings

Maintenance

Status

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 DO’s

Installation DONT’s

Applications

Overview

Interfaces

Port Configuration






Advance Settings

Maintenance

Status

LED Indications


Broadband Modem/Router

FXO1…FXO16

PSTN Network

2001

2016

FXS1…FXS16

IP Network

Ethernet

Stand Alone Call Possibilities

1. FXS IP Network

2. FXS PSTN Network

Stand Alone Application


FXO1…FXO16

PSTN Network

FXS1…FXS16

IP Network

Ethernet FXS ports of SETU VFXTH are

connected to FXO Ports of PBX. Extensions of PBX thus

can avail the PSTN & VoIP Networks of SETU VFXTH

PBX

FXS1…FXSN

SETU VFXT1616

In Front of PBX

Application

Behind the PBX Application


FXO1…FXO16

IP Network

Ethernet FXO ports of SETU VFXTH are connected to FXS

Ports of PBX. Extensions of SETU VFXTH

thus can use the Trunk of SETU VFXTH

PBX

FXS1…FXSN

2001

2016

FXS1…FXS16

PSTN N/W

SETU VFXT1616

Analog Extension PBX Over IP Application

Ethernet PBX FXS1…FXSN

2001

2016

PSTN N/W

SETU VFXT1616

FXO1… FXO16

FXO1… FXO16

Eth

ern

et

2001

FXS1… FXS16

FXO1… FXO16

SETU VFXT1616

IP Network Broadband

Modem/Router


Peer to Peer Calling

Ethernet

2001

2001

FXS1… FXS16

FXO1… FXO16

FXS1… FXS16

Ethernet

FXO1… FXO16

2016

IP Network

SETU VFXT1616

SETU VFXT1616

PSTN Call over IP (Long Distance converted to Local Call)

Ethernet

2001

2001

FXS1… FXS16

FXO1… FXO16

FXS1… FXS16

Ethernet

FXO1… FXO16

2016

IP Network

SETU VFXT1616

SETU VFXT1616

Mumbai Delhi

Programming Using Phone

Certain parameters of SETU VFXTH can be configured by dialing system commands

from a telephone connected to the FXS port

You can configure certain network parameters like IP address, Subnet Mask,

Connection Type, set the system to default and also view current IP address, Subnet

Mask, Connection Type, DNS and Gateway address by dialing system commands

Programming Using Phone

SE Login

Connect Analog Phone to FXS port of SETU VFXTH

OFF - Hook the phone

Hear Dial Tone [Toooooooooooooooooooooo]

Dial Command “#19 – SE 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)

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

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)

Commands

programming Using pc

Web Jeeves Login from Local Network

Network Switch

192.168.50.200

192.168.50.33

SETU VFXTH is located on Local IP

Internet 203.88.123.231

SETU VFXTH is located on Global IP

PC with internet connection

Web Jeeves Login from Public Network

Internet

WAN: 203.88.123.231:80

PC with internet connection

IP : 192.168.1.151 Subnet : 255.255.255.0 Gateway : 192.168.1.1

LAN: 192.168.1.1

Router’s port:80 is forwarded to IP Address of

SETU VFXTH

Web Jeeves Login from Public Network

Programming

Built – in Web server

GUI based software called Jeeves

Accessible using any web browser

Default IP of Ethernet Port is 192.168.001.136

Default SE password is 1234

Programming

Enter Ethernet Port IP Address of SETU VFXTH

Programming

Enter Password for Login (Default: 1234)

Login Page

Home Page

Basic Settings

Network Port Parameters

Select Region of system installation and accordingly Call progress tone country wise

• This parameters can be programmed as per existing data network

• Connection type :

1. Static: IP address, Subnet mask & Gateway Address assigned 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



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

Login Password

Password for Jeeves/FTP/Telnet can be minimum of 4 characters and Maximum of 16 characters long

All ASCII characters are allowed except white space & ( ) ; “ ‘ < > | \ dot (.)

Date – Time Settings

Click on arrow to Set date and time manually

Set SNTP server address here to sync date & time with SNTP server

MWI (Message Wait Indication on SIP Trunk

If you have subscribe for MWI on SIP trunk for the voice mail service by your ITSP then Program Message retrieval number provided by ITSP and port number ion which MWI is to be sent

MWI (Message Wait Indication on SIP Trunk

Incoming call management - SIP trunk - FXO Port

• The process of routing calls originated on 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

Incoming Call Route

Outgoing on which Number

Destination Number determination

FXO Port

Incoming Call Route options on FXO Port

Destination Number Determination on FXO Port

2 different routings defined here 1. Route all IC

calls (with CLI)

2. Route all IC calls (without CLI)

Without Any Destination Number

Define destination port for routing calls

• Without any Destination Number

• To the Fixed Destination Number

• On the basis of Calling Party Number

• After answering the call and collecting the digits

Destination Number Determination on FXO Port

• Incoming call on the FXO port

• All calls received on the FXO port are directly routed to the fixed destination

port, configured for this port, regardless of the destination number


FXO

022 2631725

SETU VFXTH


2001

FXS

No destination number will be provided, only Destination port will be applied


• Call is routed to the Fixed destination number programmed on that particular

trunk line using the Destination port programmed for that trunk

• Destination port can be FXS port, FXO port or SIP Trunk

Route To a Fixed Destination Number

FXO

Fixed Destination Number: 471

SIP

471@matrix- pbx.dynalias.org

0265 2630555


SETU VFXTH


• Calls are routed to a specific number according to the calling party number

• When there is an incoming call on the FXO port, SETU VFXTH will match the

calling party number with the entries of the calling party number based table,

if a match is found, the call is routed to the destination number

Route on the basis of Calling Party Number

FXO SIP


Calling Number Destination Number

02652630555 471

02226471110 472

SETU VFXTH


0265 2630555


• Incoming calls are answered and dial tone is played to the caller, allowing the

caller to dial the desired number

• The number dialed by the caller is considered as the destination number and

dial it out using the destination port programmed

After Answering the call & collecting the digits

FXO

After Answering the call & collecting the digits

SETU VFXTH

Dial Tone

SIP

471


0265 2630555

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SIP Trunk

Incoming Call Route options on SIP trunk

Destination Number Determination on SIP Trunk

Destination Port Options on SIP trunk

Destination Port Determination on SIP Trunk

2 different routings defined here 1. Route all IC

calls (with CLI)

2. Route all IC calls (without CLI)

• Without any Destination Number

• To a Fixed Destination Number

• On the basis of Calling Party Number

• To the Called Party Number

Destination Number Determination on SIP Trunk

• Incoming call on the SIP Trunk

• All calls received on the SIP Trunk are directly routed to the fixed destination

port, configured for this port, regardless of the destination number


SIP

022 2631725

SETU VFXTH

2001

FXS


No destination number will be provided, only Destination port will be applied


• Calls are routed to the Fixed destination number programmed on that SIP trunk

using the Destination port programmed for that SIP trunk

• Destination port can be FXS port, FXO port or SIP Trunk

Route to a Fixed Destination Number

SIP

Fixed Destination Number: 0265 2630555

FXO

0265 2630555


SETU VFXTH



• Calls are routed to a specific number according to the calling party number

• When there is an incoming call on the SIP 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


Calling Number Destination Number

471 02652630555

472 02226471110

SETU VFXTH


0265 2630555


• Incoming calls are routed to a desired number depending upon the called

number received in the SIP ID of request URI of the INVITE message

To the Called Party Number

SIP

SETU VFXTH

FXO

0265 2630555

0265 2630555

To the Called Party Number

203.88.142.221

02652630555@ 203.88.142.221

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Destination Port Determination

Outgoing By which Trunk

• 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

3. On the basis of calling party number (Not Supported on FXS Port)

Destination Port Determination

Destination Port Determination on FXS Port

Destination Port Options on FXS Port

Destination Port Determination on FXO Port

Destination Port Options on FXO Port

Destination Port Options on SIP trunk

Destination Port Determination on SIP Trunk

Outgoing Call Management - FXS Port - FXO Port - SIP trunk

FXS Port

For OG Call we can allow or block outgoing calls, enable flag to Block the Outgoing from this trunk

FXO Port

Enable flag to Block the Outgoing call from this trunk, Apply ANT with Dialed & substitute number string

SIP Trunk

Options Related to Outgoing calls through SIP trunk

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 traversal in SIP messages. STUN

server facilitates traversing through most NATs except symmetric NATs

STUN (Simple Traversal of UDPs through NATs)

Illustration of STUN

Illustration of STUN

STUN Request STUN Request

STUN Response

To:115.118.161.163:5060 Payload:115.118.161.163:5060

STUN Response

To: 192.168.50.161:5060 Payload:115.118.161.163:5060

Source:192.168.50.161:5060

Source: 115.118.161.163:5060

STUN Server

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STUN

Program the STUN Server IP Address & Port here

Select NAT type as STUN if you want to use IP address fetched using STUN

STUN

Status page will display the IP address, port number and NAT type fetched using STUN

STUN

Router Public IP Address

Port Forwarding:

Since STUN doesn’t work with symmetric NAT , as an alternative to STUN Port

Forwarding can be done in the router and Router’s Public address that is configured

can be used as Source Port IP Address

VoIP Port Parameters: Router’s 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 the system parameter page

P2P Call One Device is on Public IP and Other Device installed behind NAT

192.168.200.210

Internet

SETU VFXTH IP: 192.168.200.195 G/W : 192.168.200.210

Router separates Private and Public

Network

Private IP

Public IP

203.88.142.218

Port Forward in Router

LAN port of Router WAN 203.88.142.221

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

Router Configuration : Example

Router’s Network

Parameters

Port Forwarding:

Router’s SIP and RTP Ports are forwarded to Private IP of SETU VFXTH

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

Router Configuration : Example

• 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



Program SIP trunk mode as peer to peer for peer to peer calling

Enable SIP trunk


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

Proxy Calling

Requirement for Proxy Calling

Proxy server authenticates the clients for outgoing calls through it

What is required for

authentication?

SIP ID

Authentication ID

Authentication Password

Registrar Server Address

Registrar Server port

Proxy Calling

Select SIP Trunk as Proxy and assign the

authentication credentials provided by service provider

Enable the flag

Proxy Calling

If this flag is enabled, SETU VFXTH will send

the REGISTRAR MESSAGE to

Registrar Proxy as applicable

SIP Registration

On enabling the flag of SIP Registration, following parameters are to be taken care of

This is the time period after which system will send registration request

to maintain binding with Registrar Server. Valid range: 00001-65535.

Default:3600 Seconds

When a registration attempt fails, system resends request to registrar server after this timer’s expiry. Valid

range: 00001-65535. Default:10 Seconds

SIP Registration

System will get unregistered with the current server & will

register with the alternate server, if fallback occurs while sending INVITE message when

Switch Registration to Alternate Server on Fallback is

enabled

Registrar Settings

If you want the system to send

DNS SRV query to the configured domain server, enable this flag

What is DNS SRV?

Dialing by domain names lets a SIP user have a single public “SIP Address” which

can be redirected at will to their current location.

SRV records maintain stability and also opens up the possibility to use your own

domain, regardless of the domain of the SIP service you are using

SIP Registration

Enable the flag, if your service

provider supports multiple servers

in its network

Advance Settings

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

• SETU VFXTH users can access the features and facilities by dialing the access

code assigned to them from a phone. User can

1. Enable/Disable a feature

2. Access Supplementary feature

3. Enter into the programming mode

• SETU VFXTH provides default access code for all features, you can change it to

suit your preferences

Access Codes

Access Codes

Access codes can be changed from here

• 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 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 Numbers

Allowed – Denied Logic on FXS Port

Apply allowed denied list on FXS Port & program the number list for allowed & denied numbers

Apply allowed denied list on FXO port & program the number list for allowed & denied numbers

Allowed – Denied Logic on FXO Port

Allowed – Denied Logic on SIP Trunk

Apply allowed denied list on SIP trunk & program the number list for allowed & denied numbers

• This feature is used to translate the dialed number string to preprogrammed

number string

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

string

• For this feature we need to configure dialed number string and substitute

number string in number list table

• ANT feature is applied on destination ports (On all SIP trunks and FXO Ports)

Automatic Number Translation

Apply ANT on FXO port and program the table number



Apply ANT on SIP Trunk and program the table number

ANT table


Examples on how to program

• SETU VFXTH supports feature ‘Black listed Callers’ which enables you to block

incoming calls from specific numbers and addresses on the SIP trunks

• This feature is applicable on source port only

• To use this feature, user must configure the numbers of unwanted callers in a

number list

• Enable the Reject Calls from Blacklisted Caller check box on the SIP trunks on

which you want to apply this feature

Black Listed Callers

Black Listed Callers

Apply black listed caller feature on selected SIP trunk and define the number list for the same Black Listed Callers

• It’s a record for the calls, containing information about the gateway’s usage

when call was made

• Maximum of 2000 call record entries can be stored

• Call record entries are stored in FIFO logic

• User can set different filters as required and generate Call Detail Record (CDR)

report

• Call records can be cleared manually at any time

Call Detail Record (CDR)

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

1. Date of call origination

2. Time of call origination

3. Calling number

4. Called number

5. Duration of call

6. Source port

7. Destination port

8. Disconnected by

9. Cause

10. PIN number

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

6. Calls of certain called party numbers

7. Calls of certain calling party numbers

8. Calls made with PIN authentication

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

CDR can also be viewed from Jeeves


• 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 authentication is enabled on source port, SETU VFXTH answers the

Incoming call and plays a feature tone, it waits for the caller to dial the PIN

number and password, 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

attempts, if the caller fails to dial correct PIN and password in all attempts, the

system disconnects the call

PIN Authentication

PIN Authentication – FXO Port

Select routing type ‘after answering the call and collecting the digits’ for PIN authentication feature to use

Enable this flag for prompting caller to enter PIN

PIN Authentication

Enter PIN number & PIN password, system checks PIN entered by the caller during call with the entries in the PIN authentication table, if match found then only the call will be processed further

• 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 Password – in

the INVITE message are authenticated and accepted by the recipient, the

originator and recipient are connected

• You may use the digest authentication to restrict access to SETU VFXTH to

specific callers, prevent unwanted or malicious calls

Digest Authentication

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

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

2. When the calling party sends its credentials, SETU VFXTH authenticates the

credentials by matching it with its Digest Authentication table

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 Table is required when you 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

Static Routing

• 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, SETU VFXTH routes the call to the IP network using the SIP trunk

determined by the routing mechanism. The FXS user can dial only numbers not

domain names, therefore it becomes necessary that the domain names be

assigned prefix codes which the FXS user can dial

Prefix to Domain Name Conversion

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

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

some FXS port has set call forward and only number is programmed or user is

doing blind transfer

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

abc.com and destination number for call forward is *1239974 then it will be

replaced by [email protected]


Define prefix and domain name in the table


• 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

Disconnect Tone

Program the disconnect tone cadence here

• SETU VFXTH supports dialing of emergency 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, No emergency numbers are loaded in the system, in the emergency

number table

Emergency Numbers

Click here to add new entry to the table Click here to delete

entry from the table

Click here to Edit entry of the table

Emergency Numbers

Features

• 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 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

Class Of Service

Enable the supplementary services after enabling the feature in COS

Supplementary Services

• When SETU VFXTH is interfaced with service provider 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 port connected to the system

Subscriber Type

• Select Network if you want to use supplementary services supported by the

other PBX, you can access the service provider features by dialing FLASH, you

will not be able to access the local features of SETU VFXTH

• Select Gateway if you want to use supplementary services supported by the

SETU VFXTH, in the gateway mode 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

Subscriber Type

Signaling Loop 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 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 3

DTMF Detection 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 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

First Digit & Inter Digit wait timer

FXS Port

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)

FXS

Return Loss >18 dB


Transmission Level Adjust Tx Gain: -15 dB to +10 dB

Rx Gain: -15 dB to +10 dB

Call Maturity Delay & Polarity Reversal

Answer Supervision on FXO Battery Reversal

Disconnect Supervision on FXO Battery Reversal & Open Loop Disconnect

FXO Port

REN 3

DTMF Detection ITU-T Q.24

CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore

Protection Over Voltage Secondary Protection

Return Loss >18 dB


Transmission Level Adjust 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

FXO Port

Hardware settings on FXO port

FXO Port

FXO Port

General settings on FXO port

• This feature enables callers to disconnect the current call and make a new call

using SETU VFXTH without getting disconnected from 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 destination number

determination method

• If you have enabled Connect source port when number is out dialed on the FXO

port, you will not be able to provide this feature to callers

Making a new call using access code

• To make a new call using access code

In speech with the current call

Dial #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 dial #91 again to make another new call


Enable the flag to allow user making new 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 port

engaged in the call

• This feature is applicable only when destination number determination method

is selected as After answering the call and collecting digits

• If you have enabled Connect source port when number is out dialed on the

FXO port or have enabled Connect source port when 183 is received on SIP on

the SIP trunk, you will not be able to provide this feature to users

Disconnecting a call using access code


Enable the flag to allow call disconnection using access code

• 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 from the

Phone at FXS

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

does not check the destination port determination method you have

configured for that 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 for IP dialing

IP Dialing

100rel and SIP PRACK

SIP PRACK (SIP Provisional Acknowledgement) is a method to enable reliability for SIP 1XX messages

The Called Party answers the PRACK by 200 OK and PRACK is only for 1XX

messages other than 100 Trying

Generally PRACK message flows from Calling Party to Called Party

System Parameters

Enable Provisional acknowledgement for all 1xx messages other then 100 trying

SIP Timers

• SIP Invite Timer

• SIP Provisional Timer

• General Request Timer

SIP Invite Timer

• It is the time for which SETU VFXTH waits for a response from the called party

after sending INVITE message

• This time starts after sending INVITE message to the called party and stops on

receipt of provisional response or final response or when the user goes ON-

Hook, on expiry of the timer the call is disconnected

• The range of SIP INVITE Timer is 10 - 80 seconds (Default: 30 Seconds)

SIP Provisional Timer

• It is the time for which SETU 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 of SIP Provisional Timer is 10 - 180 seconds (Default: 60 Seconds)

• It is the time for which SETU VFXTH waits for the response 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 SETU VFXTH clears the transaction

• The range of SIP Provisional Timer is 10 - 60 seconds (Default: 20 Seconds)

General Request Timer

System Parameters

Program the timer values according to the requirement

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 Peer to Peer

• By Default SIP messages transported over TCP

• Disable the flag to send SIP messages over UDP

SIP Over TLS

• The SIP over TCP option allows you to send/receive the SIP messages over TLS.

TLS protects SIP signaling against loss of integrity, confidentiality and against reply

• SIP over TLS is applicable for both Proxy and Peer to Peer

• By Default SIP over TLS is enabled

• Disable the flag to disable SIP over TLS

System Parameters

Program SIP TCP, UDP and TLS port values. Default: 5060 for TCP, UDP and 5061 for TLS

Server Port

Server ports can be changed to any value from 1021 to 65,535

Management/Security

Server ports can be changed to any value from 1021 to 65,535

Certificate

Certificate

• SETU VFXTH supports certification for TLS, Web Server, Firmware Upgrade,

Configuration Upgrade and TR-069.

• SETU VFXTH supports two types of Certificates: Self-Signed Certificate and CA

Signed Certificate.

Self – Signed Certificate

• A self-signed certificate is created by the clients themselves or by the Servers and

then given to their clients.

• It means that you yourself become the Certificate Authority (CA), create a CA

Certificate and sign it.

• The self-signed certificate is faster to create but is not signed by a trusted CA

Organization.

• The self-signed certificate must be installed in the trusted list of clients that

connects over TLS with the Server. Because the certificate has been self signed, the

signature is not likely to be in the clients’ trust file, hence, they need to add it.

Certificate

Generate self signed CA certificate by entering the required details below

Once you generate self-signed certificate, you must send it to your clients so that they install it in their trusted list.

Click generate to generate new certificate for entered details

Certificate

System will show generated certificates under trusted root CA

System Certificate

• After creating a Self-Signed CA Certificate, you can either,

• Generate a System Certificate for your clients. These System Certificates can then be given to the respective clients OR

• The Clients can prepare their own System Certificates. For this you need to send them the CA Certificate created by you OR

• Generate a Certificate Signing Request (CSR), if you want the Certificate to be signed by a third party

If the clients prepare their own certificates, you need to send your CA Certificate to all the clients. The clients must upload the same in their system. Similarly, all the clients must send their CA Certificates to you and you must upload the same in your system. To avoid this, it is recommended that you create the Certificates and then provide it to your clients

Enter details to generate system certificate

If you want to get a CA Signed Certificate, you need to do the following: 1. Generate and enroll the Certificate Signing Request (CSR). 2. Get the Certificate Signing Request (CSR) verified and signed by the Certified Authority (CA).

Certificate

List of available system certificates

User can also upload the certificates

Certificate

Define the certificate to be used for desired application

Maintenance

Firmware

Browse the ZIP file having new firmware files & click on Upgrade button to upgrade the system firmware

Program the details for Auto firmware upgrade

Upgrade firmware automatically from Matrix Server

Configuration

Browse the ZIP file having configuration files & click on Upgrade button to upgrade the system configuration

Program the details for Auto configuration

Click on Backup Configuration to save config.zip file

• 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, Syslog client enables the

system to send debug messages in Syslog format to the remote ‘Syslog server’

on the IP network

• Syslog uses the UDP as transport protocol

• To be able to use this feature, you must enable ‘Syslog’, configure the Syslog

Server Address and define the server port on which the Syslog will listen for

debug messages

System Debug

System Debug

Debug events can be viewed on the screen

Click debug settings to set parameters for debug and to start debug in PC/Laptop connected to SETU VFXTH

System Debug

Program the IP address and port number of PC/Laptop where Syslog server is installed

Debug for Port: clear the check box to disable the debug for the port which is not needed

• SNMP – Simple Network Management Protocol

• SNMP protocols supported – SNMPV1, SNMPV2C, SNMPV3

• SETU VFXTH is having built in SNMP Server (SNMP Server). It receives SNMP

requests and generates SNMP responses or notifications

• SNMP Manager usually network management station. It generates SNMP

requests and receives SNMP responses and notifications. The SNMP manager is

an SNMP client

SNMP

SNMP

Program SNMP details

System Port Activity

System port activity like Idle, Inactive, Disable, Dial, Speech, ringing, Incoming Call Proceeding, Remote Held, Error

• 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

PCAP Trace

Click here to start the PCAP trace Click here to stop

the PCAP trace

Once the PCAP is captured save the trace file on your PC/Laptop

Click here to Enable Promiscuous mode

Enter the filter details here

• Select source port and destination port with source number and destination

number.

• When Call button from GUI is pressed system will call source number first and

when answered by source port it will ring on destination port & speech path

can be checked

• Clicking on call button will also lead the programmer to system port activity

page to monitor the status of the port during call progress

Manual Call Test

Manual Call Test

• SETU VFXTH supports the AC Impedance Test for clear, audible and echo-free

speech over FXO Ports.

• This test helps you to set the most appropriate values for the FXO Port

Parameters —AC Termination Impedance, CO Termination and CO Line Type— to

correct the line impedance mismatch between the AC Termination Impedance

presented by the FXO Port of SETU VFXTH to the line and the CO Termination

Impedance presented by the Central Office to the line.

• While the test is being conducted, you will hear pulsating tone on all the ports of

the system. (Mute the microphone of destination landline number or mobile

number when call is answered by destination number)

AC Impedance Test (FXO)

AC Impedance Test (FXO) Enter phone number on which system will make the call in order to complete the test

Select the FXO port on which you want to run the test

Click on start test and wait for the results

For more details click help

Default System

Click OK to factory default the gateway

Soft Restart

Click OK to Restart SETU VFXTH

• TR-069, also known as CPE WAN Management Protocol (CWMP), is a remote

management protocol used for secure communication between the Customer

Premises Equipment (CPE) and an Auto-Configuration Server (ACS) for various

functionalities such as:

Auto-configuration and dynamic service provisioning

Firmware Management

Status and performance monitoring

Diagnostics

• SETU VFXTH supports TR-069. Using TR-069, service providers can manage SETU

VFXTH remotely for the functions described above.

TR – 069

TR – 069

Program TR-069 details

STATUS

System Detail

Version Revision details

Firmware Status

Last Firmware up gradation details if scheduled firmware upgrade is ON

Configuration Status

Last Configuration up gradation details if scheduled firmware upgrade is ON

Network Status

IP details status of IP configured in SETU VFXTH

FXO Port Status

Line connection status on FXO

SIP Trunk Status

SIP trunk Status

SETU VFXTH March 2014

Technology

Transcript of SETU VFXTH March 2014