6. • 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
14. LED Indication – PWR
LED Status Related Information
PWR ON Power is ON
OFF Power is OFF
15. 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)
16. 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
17. 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
18. 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 (LED L1 Indication)
19. LED Indication – T1E1 Port
System Status Colour Cadence
No Alarm Green Continuous ON
RAI Alarm Red 500 ms ON – 500 ms OFF
AIS or LOS Alarm Red Continuous ON
Signaling Type: E1 PRI (LED L2 Indication)
20. LED Indication – T1E1 Port
System Status Colour Cadence
No Alarm Green Continuous ON
CRC4 Alarm Green 100 ms ON – 100 ms OFF
MFA Alarm Red 100 ms ON – 100 ms OFF
BFA Alarm Red 500 ms ON – 500 ms OFF
LOS Alarm Red Continuous ON
Signaling Type: E1 CAS (LED L1 Indication)
21. LED Indication – T1E1 Port
System Status Colour Cadence
No Alarm Green Continuous ON
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)
22. LED Indication – T1E1 Port
System Status Colour Cadence
No Alarm Green Continuous ON
TFA Alarm or MFA Alarm Red 500 ms ON – 500 ms OFF
AIS Alarm Red 100 ms ON – 100 ms OFF
LOS Alarm Red Continuous ON
Signaling Type: T1 RBS or T1 – PRI (LED L1 Indication)
23. LED Indication – T1E1 Port
System Status Colour Cadence
No RAI Alarm Green Continuous ON
RAI or LOS Alarm Red Continuous ON
Signaling Type: T1 RBS or T1 – PRI (LED L2 Indication)
33. 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
34. 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
35. 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
36. 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
41. 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
44. 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
48. 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
49. 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
50. Fixed Destination Number: 9662043785
9662043785
SIP1
Fixed Destination Number: 8271110
SIP2
T1E1
SIP10
8271110@matrix-
pbx.dynalias.org
5496767@
iptel.org
468@matrix-
pbx.dynalias.org
Route To a Fixed Destination Number
51. Route To a Fixed Destination Number
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
52. Incoming call on the SIP trunk
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
54. Route on the basis of Calling
Party Number
Select route for
all Incoming calls
as on the basis of
calling party
number
2 different routings
defined here
1. Route all IC calls
(with CLI)
2. Route all IC calls
(without CLI)
55. Route on the basis of Calling
Party Number
Program the calling
number based table with
calling party number and
destination number
56. Incoming call on the SIP trunk
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
58. Route to the Called Party Number
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)
59. Incoming call on the SIP trunk
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
61. After Answering the call &
collecting the digits
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
62. Incoming call on the SIP trunk
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
63. SIP
9974098915
On the basis of DDI Number
2001@matrix-
pbx.dynalias.org
SETU VTEP
DDI Number Destination Number
662501 2001
662501 2002
192.168.50.123
64. On the basis of DDI Number
Define destination
port for routing calls
2 different routings defined here
1. Route all IC calls (with CLI)
2. Route all IC calls (without CLI)
65. On the basis of DDI Number
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
66. On the basis of DDI Number
Enter the details required for DDI
numbers generation
68. 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
T1E1 Trunk “Port wise Routing”
70. 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
Route To a Fixed Destination Number
71. Fixed Destination Number: 5111471
5111471@
iptel.org
SIP1
Fixed Destination Number: 8271110
SIP2
T1E1
8271110@matrix-
pbx.dynalias.org
812868304
2
9974098915
Route To a Fixed Destination Number
T1E1
72. Route To a Fixed Destination Number
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
Define destination
port for routing calls
73. Incoming call on the T1/E1 port
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
Route on the basis of Calling
Party Number
75. Route on the basis of Calling Party
Number
Select route for all Incoming
calls as on the basis of calling
party number
2 different routings defined here
1. Route all IC calls (with CLI)
2. Route all IC calls (without CLI)
76. Route on the basis of Calling Party
Number
Program the calling
number based table with
calling party number and
destination number
77. Incoming call on the T1/E1 port
Call is routed to a specific number depending upon the called party number received
in the SETUP message of the T1/E1 port
Route to the Called Party Number
78. Route to the Called Party Number
3301@matrix-
pbx.dynalias.org
SETU VTEP
ISDN PBX
2001
3301
SIP T1E1
79. Route to the Called Party Number
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
80. Incoming call on the T1/E1 port
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
After Answering the call &
collecting the digits
82. After Answering the call &
collecting the digits
Select route for all Incoming
calls as after answering the call
and collecting the digits
2 different routings defined here
1. Route all IC calls (with CLI)
2. Route all IC calls (without CLI)
Define destination
port for routing calls
83. Incoming call on the T1/E1 port
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
On the basis of DDI Number
84. SIP
9974098915
On the basis of DDI Number
2001@matrix-
pbx.dynalias.org
SETU VTEP
T1E1
DDI Number Destination Number
662501 2001
662501 2002
85. On the basis of DDI 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
86. On the basis of DDI Number
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
87. On the basis of DDI Number
Enter the details required for
DDI numbers generation
89. 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
91. 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
93. 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
95. Fixed
Program the routing group
for routing of Incoming calls
on the SIP trunk
Click on Edit to
change the members
of routing group
96. 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
98. 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
99. 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
101. 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
102. 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
104. 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
105. 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
107. 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
110. 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
111. 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
112. 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
113. 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
115. Making an IP call without the intervention of a proxy server is called peer to peer
calling
As peer to peer calling does not require a proxy server, voice communication using
this application can be done virtually free of cost
The major cost savings offered by this application makes it a very attractive mode
of inter – branch or intra – office voice communication
Peer to Peer Calling
116. Peer to Peer Calling
Program SIP trunk
mode as peer to peer
for peer to peer calling
Enable
SIP trunk
117. Peer to Peer Calling
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
119. 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
120. Proxy Calling
Select SIP Trunk as Proxy
and assign the
authentication
credentials provided by
service provider
Enable
the flag
121. Proxy Calling
If this flag is enabled,
SETU VTEP will send
the REGISTRAR
MESSAGE to
Registrar Proxy as
applicable
122. 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
123. 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
124. Registrar Settings
If you want the
system to send DNS
SRV query to the
configured domain
server, enable this
flag
126. 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
127. Check SIP ID for Incoming SIP Message-
Flag is Enabled
3301@matrix-
pbx.dynalias.org
128. Check SIP ID for Incoming SIP Message-
Flag is Disabled
3301@matrix-
pbx.dynalias.org
129. 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
136. 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)
137. 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
138. 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
142. 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
144. Router Public IP Address
Use NAT type as Router
Public IP address
145. Router Public IP Address
Status page will display
the Router Public IP
address programmed in
the system parameter
page
146. 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
147. *Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router’s Network
Parameters
Router Configuration: Example
148. 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.
150. 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
152. 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
153. Allowed – Denied Numbers
Apply allowed denied list on SIP
trunk & program the list number
for allowed & denied numbers
154. Allowed – Denied Numbers
Apply allowed denied list on
T1/E1 port& program the list
number for allowed & denied
numbers
155. Allowed – Denied Numbers
Program the digits for allowed &
denied numbers in number list
table
156. 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
157. 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
158. Automatic Number Translation
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
160. 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
161. Black Listed Callers
Apply black listed caller
feature on selected SIP
trunk and define the
number list for the same
163. 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)
164. 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
Call Detail Record (CDR)
165. 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
Call Detail Record (CDR)
166. Call Detail Record (CDR)
Set filter parameters
for CDR here
Click here to clear
all call records
167. Click on download to get
Zip file containing CDR
in .csv and .txt format
Call Detail Record (CDR)
Save Zip file & extract it to get
CDR in .csv and .txt file
169. 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
170. 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
171. 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
172. 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
173. 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
174. 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
175. 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
Digest Authentication
178. 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
179. 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
180. 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
181. 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
183. 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.
184. 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.
185. 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
187. 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
188. 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).
192. 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
193. 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
Making a new call using access code
194. Making a new call using access code
Enable the flag to allow
user making new call
using access code
195. Making a new call using access code
Enable the flag to allow
user making new call
using access code
196. 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
199. 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
201. 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
204. 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
205. 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
206. 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
207. 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
208. 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
209. 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
211. System Port Activity
System port activity
like Idle, Inactive,
Disable, Dial, Speech,
ringing, Incoming Call
Proceeding, Remote
Held, Error
212. 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
213. 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
214. 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
218. 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
219. • 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
