SETU VTEP March 2014

Hearty Welcome!

Technical Training

SETU VTEP

• Overview

• Interfaces

• Port Configuration

• Hardware Architecture

• LED Indications

• Installation Do’s & Don'ts

Agenda

Applications

Accessing the JEEVES

Routing Scenarios

Advance Setting Parameters

Maintenance

Status

Agenda

Overview

• VoIP – ISDN PRI Gateway

• VoIP access device for an existing PBX

• PRI trunking for an IP – PBX

• Software configurable T1 and E1 PRI

• DDI routing over IP

• Dedicated Sync – In and Sync – Out port

SETU VTEP Overview

Interfaces

Interfaces

Laptop

Adaptor (5VDC = 2Amax)

Router

Switch

ISDN PRI

Sync IN

Sync OUT

Network Clock Synchronization

Port configuration

Port Configuration

Resources Capacity

USB Port (Future Use) 1

Ethernet Port 1

T1E1 Port 1

DC Power Input Jack 1

Sync In Port 1

Sync Out Port 1

Hardware Architecture

SETU VTEP Overview

LED Indication

LED Indication – PWR

LED Status Related Information

PWR ON Power is ON

OFF Power is OFF

LED Indication – Reset Sequence

System Status Colour L1 L2 Status Time in msec

Application Load Green ON OFF ON 200ms

All init done, system goes

live

Green ON ON ON 1000ms

Red ON ON ON 1000ms

OFF OFF OFF 1000ms

Green OFF ON OFF 1000ms

OFF OFF OFF 1000ms (Continuous

Last 2 steps)

LED Indication – SIP Trunk

System Status Colour Cadence (1 Cadence is of 4000ms)

ON OFF ON OFF

SIP Disable OFF

SIP Registered (Active) Green Continuous

SIP Registration Failed Red Continuous

SIP Authentication Failed Red 200

200

200

3400

LED Indication – T1E1 Port

System Status Colour Cadence

No Alarm Green Continuous ON

CRC4 Alarm Green 100 ms ON – 100 ms OFF

BFA Alarm Red 500 ms ON – 500 ms OFF

LOS Alarm Red Continuous ON

Signaling Type: E1 PRI







Signaling Type: E1 PRI (LED L1 Indication)




RAI Alarm Red 500 ms ON – 500 ms OFF

AIS or LOS Alarm Red Continuous ON

Signaling Type: E1 PRI (LED L2 Indication)





MFA Alarm Red 100 ms ON – 100 ms OFF



Signaling Type: E1 CAS (LED L1 Indication)




Y – Bit Alarm Green 100 ms ON – 100 ms OFF

AIS 16 Alarm Red 100 ms ON – 100 ms OFF

RAI Alarm Red 500 ms ON – 500 ms OFF

AIS or LOS Alarm Red Continuous ON

Signaling Type: E1 CAS (LED L2 Indication)




TFA Alarm or MFA Alarm Red 500 ms ON – 500 ms OFF

AIS Alarm Red 100 ms ON – 100 ms OFF


Signaling Type: T1 RBS or T1 – PRI (LED L1 Indication)



No RAI Alarm Green Continuous ON

RAI or LOS Alarm Red Continuous ON

Signaling Type: T1 RBS or T1 – PRI (LED L2 Indication)

Installation Do’s &don’ts

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 Don’ts

Applications

VoIP Access device for existing PBX Application

PRI Gateway for an IP – PBX

Peer to Peer & Proxy Calling

Virtual Trunking

Programming Using pc

SETU VTEP : Configuration

Web Jeeves Login from Local Network

Network Switch

192.168.50.200

192.168.50.33

SETU VTEP is located on Local IP

Web Jeeves Login from Public Network

Internet

SETU VTEP : Configuration

203.88.123.231

SETU VTEP is located on Global IP

PC with internet connection

Web Jeeves Login from Public Network

Internet

SETU VTEP: Configuration

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 VTEP

Built – in Web server

GUI based software called JEEVES

Accessible using any web browser

Default IP of Ethernet Port is 192.168.001.100

Default SE password is 1234

Programming

Click on Start Internet Explorer (Any Browser)

Programming

Enter Ethernet Port IP Address of SETU VTEP

Login Page

Enter SE Password for Login (Default: 1234)

Home Page

This parameter 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

Network Port Parameters

Network Port Parameters

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

Incoming call management

SIP trunk & T1E1 trunk

The process of routing calls originated on T1/E1 port and SIP trunks to the

destination port in SETU VTEP takes place in two steps:

1. Determination of destination number

2. Determination of destination port

Incoming Call Route

Destination Number determination

SIP Trunk

Destination Number Determination on SIP Trunk

Incoming Call Route options on SIP Trunk

To a Fixed Destination Number

On the basis of Calling Party Number

On the basis of DDI Number

To the Called Party Number

After answering the call and collecting the digits

Destination Number Determination on SIP Trunk

Incoming call on the SIP trunk

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 SIP or T1E1

Route To a Fixed Destination Number

Fixed Destination Number: 9662043785

9662043785

SIP1


SIP2

T1E1

SIP10

8271110@matrix- pbx.dynalias.org

5496767@ iptel.org



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2 different routings defined here 1. Route all IC calls

(with CLI) 2. Route all IC calls

(without CLI)

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

Enable this flag if you want to block the IC calls received without CLI on SIP trunk

Define destination port for routing calls


Call is routed to a specific number according to the calling party number

When there is an incoming call on the SIP trunk, SETU VTEP 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

9974044583

3301 T1E1



Calling Number Destination Number

3301 9974044583

8471110 8128683042


Select route for all Incoming calls as on the basis of calling party number

2 different routings defined here 1. Route all IC calls

(with CLI) 2. Route all IC calls

(without CLI)


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


Call is routed to a specific number depending upon the called party number received

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

Route to the Called Party Number

96620 43785

192.168.50.123


9662043785

SETU VTEP

192.168.50.33


Select route for all Incoming calls as to the called party number

2 different routings defined here 1. Route all IC calls (with

CLI) 2. Route all IC calls

(without CLI)


Incoming call is 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

SIP



9974099740

SETU VTEP

Dial Tone

T1E1

9974099740



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

CLI)

Select route for all Incoming calls as after answering the call and collecting the digits


A call is routed to a specific number as per the DDI number received in the SETUP

message of the SIP trunk

The DDI number based table is referred for the same and call is routed to the

destination number programmed opposite to the DDI number field

On the Basis of DDI Number

SIP

9974098915



SETU VTEP

DDI Number Destination Number

662501 2001

662501 2002

192.168.50.123




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


Program the DDI number table with DDI number and destination number

Apply Reverse DDI flag if you want to display DDI number for respective DDI number

Click here to generate sequence of DDI numbers


Enter the details required for DDI numbers generation

T1E1 Trunk “Port Wise routing”

To a Fixed Destination Number

On the basis of Calling Party Number


To the Called Party Number

After answering the call and collecting the digits

Destination Number Determination on T1E1 Trunk “Port wise Routing”

T1E1 Trunk IC

Incoming Call Route options on T1E1 Trunk

Incoming call on the T1/E1 port

Call is routed to the Fixed destination number programmed on the T1/E1 port

Destination port can be SIP or T1/E1



5111471@ iptel.org

SIP1


SIP2

T1E1


8128683042

9974098915


T1E1




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



Call is routed to a specific number, as per the calling party’s number

When there is an incoming call on the SIP trunk, SETU VTEP 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


3301



Calling Number Destination Number

9974044583 3301

8128683042 8471110

T1E1

9974044583


Select route for all Incoming calls as on the basis of calling party number



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


Call is routed to a specific number depending upon the called party number received

in the SETUP message of the T1/E1 port




SETU VTEP

ISDN PBX

2001

3301

SIP T1E1


Select route for all Incoming calls as to the called party number

2 different routings defined here 1. Route all IC

calls (with CLI)

2. Route all IC calls (without CLI) Define destination

port for routing calls


Incoming call is 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 dialed out

using the destination port programmed


SIP

9974098915


SETU VTEP

Dial Tone

T1E1

3301




Select route for all Incoming calls as after answering the call and collecting the digits




A call is routed to a specific number as per the DDI number received in the SETUP

message of the T1/E1 port

The DDI number based table is referred for the same and call is routed to the

destination number programmed opposite to the DDI number field


SIP

9974098915


2001@matrix-

pbx.dynalias.org

SETU VTEP

T1E1

DDI Number Destination Number

662501 2001

662501 2002






Program the DDI number table with DDI number and destination number

Apply Reverse DDI flag if you want to display DDI number for respective DDI number

Click here to generate sequence of DDI numbers


Enter the details required for DDI numbers generation

T1E1 Trunk Channel Number Wise routing

Channel Number Wise Routing

Routing mechanism channel wise

Channel Number Wise Routing is used in cases like 1. Service provider has bind the channels or 2. You want the routing like call on 1st channel should go to SIP1, call on 2nd channel should go

to SIP2 and so on

T1E1 Trunk MSN/DDI Number Wise Routing

MSN/DDI Number Wise Routing

Enter the details of MSN number and program the Routing mechanism

MSN number wise routing is used when you want to set same routing for a one group of DDI

numbers, and different routing for second group of DDI numbers

Destination Port Determination

SETU VTEP supports different methods of determining the destination port for the

calls originated on SIP trunks and on the T1/E1 port

1. Fixed

2. On the basis of destination number

3. On the basis of calling party number

Destination Port Determination

Destination port determination – Fixed

Fixed

Program the routing group for routing of Incoming calls on the SIP trunk

Click on Edit to change the members of routing group

Fixed

Program Group Member for routing group

Click here to Apply fallback routing group

Fallback Routing group is used in case if all members of routing group are busy or the trunk line is down

Destination port determination – Destination Number Based

Destination Number Based

Program the Destination number and routing group in the destination number based routing table

Click on Add to add the new entry for destination number

Click on Delete to delete the selected entry for destination number

Destination Number Based

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

Program the routing group and fallback routing group for the destination number defined above

Destination Port Determination – Calling Number Based

Calling Number Based

Program the calling number and routing group in the calling number based routing table

Click on Add to add the new entry for Calling Number

Click on Delete to delete the selected entry for Calling Number

Calling Number Based

Program the routing group and fallback routing group for the Calling number defined above

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

Outgoing call management -SIP trunk -T1E1 trunk

Call Block on SIP Trunk

There is no routing for Outgoing calls needed. Either we can allow or block outgoing calls, enable flag to Block the Outgoing from this trunk

Click here to Apply ANT logic on the trunk and program the number list for ANT

Call Block on T1E1 Port

There is no routing for outgoing calls needed. Either we can allow or block outgoing calls, enable flag to Block the Outgoing from this trunk

Click here to Apply ANT logic on the trunk and program the number list for ANT

Route unconnected calls to original caller

If you enable this feature, when an outgoing call is made using this port, and the called

party is found busy or does not respond, SETU VTEP stores the number of the calling

party, the number of the called party and the port (through which the outgoing call was

made)

A record of each such call is stored for the duration of the unconnected calls record

delete timer

If the called party returns the call before the expiry of this timer, this incoming call is

placed to the original calling party

User can change the duration of this timer and delete records of such calls

RCOC

RCOC

Click here to Enable RCOC on SIP trunk

RCOC

Click here to Enable RCOC on T1E1 trunk

RCOC

Set the time for which you want the system to store the records of unconnected calls (Default: 999 Minutes)

Click here to clear the

unconnected calls record

Handling of OG Calls-SIP Trunk

Enabling this allows you to make OG calls irrespective of whether this SIP trunk is successfully registered or

not If you don’t want to send CLI,

enable this flag

To connect the source port with destination port without waiting

for the call to get matured, enable this

flag

Handling of OG Calls-SIP Trunk

If you want the system to apply reverse DDI logic through this SIP

trunk then enable this flag

On enabling this, system will send the CLI received on the source port in the

FROM field of INVITE message for an OG

call

Handling of OG Calls-T1E1 Port

To connect the source port with destination port without waiting

for the call to get matured, enable this

flag

To connect source port

with the destination

port as soon as progress

indicator is received on T1E1 port,

enable this flag

Peer to Peer Calling

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 VTEP 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

Registrar Settings

Check SIP ID for Incoming SIP Message

CASE: Suppose there is a soft PBX on which SETU VTEP is registered and is getting a proxy SIP trunk

During outgoing call from SETU VTEP, it will behave like a peer to peer call

During incoming call, if the flag of Check SIP ID for Incoming SIP Message is enabled, SETU VTEP will not allow that call from that soft PBX

If this flag is disabled, all calls from soft PBX will get passed through SETU VTEP

Check SIP ID for Incoming SIP Message- Flag is Enabled

3301@matrix-

pbx.dynalias.org

Check SIP ID for Incoming SIP Message- Flag is Disabled

3301@matrix-

pbx.dynalias.org

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

Public/Private Network


Define network type for PRI whether it is public (Service Provider PRI) or private (Tie up with other PBX)

SIP

3301@matrix-

pbx.dynalias.org

9974099740

SETU VTEP

Dial Tone

T1E1

9974099740

Network – Public Pilot Number – 2630555

CLI: 2630555



SIP

3301@matrix-

pbx.dynalias.org

SETU VTEP

Dial Tone

T1E1

2001

Network – Private CLI: 3301@matrix

– pbx.dynalias.org

2001

ISDN PBX


STUN

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

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 Address; Listening Port of STUN Server (1024-65535) Default port : 03478; Enable the Flag ‘Use SIP Port fetched using STUN’ if SIP port required to be fetched by STUN else disable when Port Forwarding in the Router is done for SIP messages

STUN

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

STUN

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

Router Public IP Address

Program Router Public IP Address here


Use NAT type as Router Public IP address


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 VTEP

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 WAN

203.88.142.221

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

Router’s Network Parameters

Router Configuration: Example

Port Forwarding: Router’s SIP an RTP Port

forwarded to Private IP of SETU VTEP

Router Configuration: Example

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

Advance Settings

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

SETU VTEP 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. Knowing the current IP address, subnet mask, gateway address and DNS

address of the system (Access code for this application applicable on the T1E1

port only )

SETU VTEP 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

Allowed – denied logic can be applied on source port SIP trunks and T1/E1 port

Total 24 lists can be programmed each having 64 entries

Allowed – denied logic is not applicable for emergency numbers, access codes and

when the method for Incoming calls is:

1. On the basis of called party number

2. To a fixed destination number

3. On the basis of DDI number

Allowed – Denied Numbers


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


Apply allowed denied list on T1/E1 port& program the list number for allowed & denied

numbers


Program the digits for allowed & denied numbers in number list

table

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

Total 24 lists can be programmed each having 64 entries

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 T1/E1 Port with

orientation type – Terminal)

Automatic Number Translation


Apply ANT on T1E1 Port and program the number list for dialed and

substitute number string and select the pause timer if you have configured ^

(pause) in the ANT table


Apply ANT on SIP trunk and program the number list for dialed and

substitute number string and select the pause timer if you have configured ^

(pause) in the ANT table


SETU VTEP 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 trunk on which you

want to apply this feature

Black Listed Callers


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


Program the number list with the CLI of 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

A call is stored when it gets matured

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 originated (Source Port)

2. The port on which the calls terminated (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


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

CDR opened with notepad


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 VTEP

PIN authentication can be used on the source port only if the incoming call routing for

the source port is set to Route calls 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 VTEP 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 VTEP 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 – SIP Trunk

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 – T1E1 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 VTEP 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 VTEP will challenge the identity of the calling party

2. When the calling party sends its credentials, SETU VTEP 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 VTEP 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 VTEP 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

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

SETU VTEP 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, 911, 112, 000, 106 emergency numbers are loaded in the system, in the

emergency number table

Emergency Numbers

Emergency Numbers

Emergency numbers with routing group

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

Certificate

Certificate

SETU VTEP supports certification for TLS, Web Server, Firmware Upgrade,

Configuration Upgrade and TR-069.

SETU VTEP 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.

Self – Signed 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

Features

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

VTEP 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 source port and only when After answering the call

and collecting digits is selected as the destination number determination method

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


Enable the flag to allow user making new call using access code

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

When the call disconnect access code is dialed, SETU VTEP 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

Disconnecting a call using access code



SETU VTEP 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 VTEP, 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

IP Dialing

SIP trunk or SIP trunk group can be defined IP dialing

You can know the current IP address, Subnet mask, Gateway address and DNS address of

SETU VTEP by dialing the specific access codes on the T1/E1 Port

• To do this call on the ISDN number of the T1/E1 port

• To know the current IP address, dial #51

• To know the Subnet mask, dial #52

• To know the current Gateway address, dial #53

• To know the current DNS address, dial #54

The system will announce the IP address, Subnet mask, Gateway address and DNS

address according to the access code you dial

Knowing Network Information using Access Code

Knowing Network Information using Access Code

Default access code to know network information

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 VTEP 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 VTEP 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 VTEP also supports filter setting can also be used to target the particular IP, Port

number etc.

If promiscuous mode is enabled, SETU VTEP will capture all network traffic and if disabled

then system will capture only traffic that is directly related to SETU VTEP (to or from SETU

VTEP)

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

Default System

Click OK to factory default the SETU VTEP

Soft Restart

Click OK to Restart SETU VTEP

T1E1 Port Alarms/Performance Monitoring

To detect errors on the T1E1 port there are:

RED alarm: Loss of signal YELLOW alarm: Remote alarm

indication BLUE alarm: Alarm Indication

Signal

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

VTEP 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 VTEP

SIP Trunk Status

SIP trunk Status

T1E1 Port Status

T1E1 Port status whether both layers are Up or Down

Technology

SETU VTEP March 2014