4. 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
10. 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
12. 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
13. 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
15. Installations Do’s and Don’ts
Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
21. Behind the PBX Application
Broadband
Modem/Router
FXO1…FXO16
IP
Network
EthernetFXO 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
22. Analog Extension PBX Over IP Application
Ethernet PBX FXS1…FXSN
2001
2016
PSTN
N/W
SETU
VFXT1616
FXO1…
FXO16
FXO1…
FXO16
Ethernet
2001
FXS1…
FXS16
FXO1…
FXO16
SETU
VFXT1616
IP
Network Broadband
Modem/Router
Broadband
Modem/Router
26. 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
27. 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
29. 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
34. 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
41. • 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
Network Port Parameters
42. Network Port Parameters
Select connection
type of SETU VFXTH
and according to the
connection type
program the IP details
43. 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 (.)
44. Date – Time Settings
Click on arrow to Set
date and time manually
Set SNTP server address
here to sync date &
time with SNTP server
45. 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
48. • 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
52. 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
53. • 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
54. • 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
Without Any Destination Number
55. FXO
022 2631725
SETU VFXTH
Without Any Destination Number
2001
FXS
No destination number will be provided, only Destination port will be applied
56. • Incoming call on the FXO port
• 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
57. FXO
Fixed Destination Number: 471
SIP
471@matrix-
pbx.dynalias.org
0265 2630555
Route To a Fixed Destination Number
SETU VFXTH
58. • Incoming call on the FXO port
• 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
59. FXO
SIP
Route on the basis of Calling Party Number
Calling Number Destination Number
02652630555 471
02226471110 472
SETU VFXTH
471@matrix-
pbx.dynalias.org
0265 2630555
60. • Incoming call on the FXO port
• 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
61. FXO
After Answering the call & collecting the digits
SETU VFXTH
Dial Tone
SIP
471
471@matrix-
pbx.dynalias.org
0265 2630555
64. 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)
65. • 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
66. • 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
Without Any Destination Number
68. • Incoming call on the SIP Trunk
• 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
69. SIP
Fixed Destination Number: 0265 2630555
FXO
0265 2630555
Route To a Fixed Destination Number
SETU VFXTH
471@matrix-
pbx.dynalias.org
70. • Incoming call on the SIP Trunk
• 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
Route on the basis of Calling Party Number
71. SIP
FXO
Route on the basis of Calling Party Number
Calling Number Destination Number
471 02652630555
472 02226471110
SETU VFXTH
471@matrix-
pbx.dynalias.org
0265 2630555
72. • Incoming call on the SIP Trunk
• 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
75. • 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
84. 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)
91. 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
92. Router Public IP Address
Use NAT type as Router
Public IP address
94. Router Public IP Address
Status page will display the
Router Public IP address
programmed in the system
parameter page
95. 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
96. *Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router Configuration : Example
Router’s
Network
Parameters
97. 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
99. • 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
100. Peer to Peer Calling
Program SIP trunk
mode as peer to peer
for peer to peer calling
Enable
SIP trunk
101. 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
103. 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
104. Proxy Calling
Select SIP Trunk as
Proxy and assign the
authentication
credentials provided
by service provider
Enable
the flag
105. Proxy Calling
If this flag is enabled,
SETU VFXTH will send
the REGISTRAR
MESSAGE to
Registrar Proxy as
applicable
106. 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
107. 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
108. Registrar Settings
If you want the
system to send
DNS SRV query to
the configured
domain server,
enable this flag
109. 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
112. • 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
115. • 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
116. Allowed – Denied Logic on FXS Port
Apply allowed denied list on FXS
Port & program the number list
for allowed & denied numbers
117. Apply allowed denied list on FXO
port & program the number list
for allowed & denied numbers
Allowed – Denied Logic on FXO Port
118. Allowed – Denied Logic on SIP Trunk
Apply allowed denied list on SIP
trunk & program the number list
for allowed & denied numbers
119. • 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
120. Apply ANT on FXO port and
program the table number
Automatic Number Translation
123. • 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
124. Black Listed Callers
Apply black listed
caller feature on
selected SIP trunk and
define the number list
for the same Black
Listed Callers
125. • 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)
126. • 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)
127. • 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
Call Detail Record (CDR)
128. Call Detail Record (CDR)
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
129. CDR can also be
viewed from Jeeves
Call Detail Record (CDR)
130. • 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
131. • 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
132. 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
133. 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
134. • 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
135. • 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
Digest Authentication
136. Enable apply flag in
SIP trunk to use
digest authentication
Digest Authentication
138. • 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
139. 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
140. • 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
141. • 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 9974@abc.com
Prefix to Domain Name Conversion
143. • 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
146. • 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
147. 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
149. • 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
152. • 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
153. • 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
155. 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
156. 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
160. 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
161. 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
162. Return Loss >18 dB
Longitudinal Balance >50 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
163. 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
FXO Port
166. • 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
167. • 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
168. Enable the flag to allow
user making new call
using access code
Making a new call using access code
169. • 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
170. Disconnecting a call using access code
Enable the flag to allow
call disconnection using
access code
171. Enable the flag to allow
call disconnection using
access code
Disconnecting a call using access code
172. • 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
173. SIP trunk or
SIP trunk
group can be
defined for IP
dialing
IP Dialing
174. 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
177. 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)
178. 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)
179. • 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
181. 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
182. 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
187. 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.
188. 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.
189. 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
191. 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
192. 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).
196. 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
197. 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
198. • 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
199. System Debug
ug events
be viewed
he screen
Click debug settings to set
parameters for debug and
to start debug in PC/Laptop
connected to SETU VFXTH
200. 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
201. • 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
203. System Port Activity
System port activity
like Idle, Inactive,
Disable, Dial, Speech,
ringing, Incoming Call
Proceeding, Remote
Held, Error
204. • 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
205. 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
206. • 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
208. • 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)
209. 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
212. • 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