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Hearty Welcome!
Technical Training
SETU VFXTH
“VoIP – FXO – FXS Gateway”
Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
A Versatile VoIP-FXO-FXS Gateway
A Gateway that provides voice service over IP network using SIP protocol
An effective and flexible solution for accessing Internet based telephone services &
corporate Internet systems across established LAN
Developed to fulfill requirements of SOHO (Small Office Home Office) users & small/
medium scale enterprises
Overview
Interfaces
Overview
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
Interfaces
Port Configuration
Overview
Interfaces
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
Configurations VoIP
Channels
FXO
Ports
FXS
Ports
FXO Ports
Label
FXS Ports
Label
SETU VFXTH0016 16 0 16 0 P01-P16
SETU VFXTH0024 24 0 24 0 P01-P24
SETU VFXTH0032 32 0 32 0 P01-P32
SETU VFXTH3200 32 32 0 P01-P32 0
SETU VFXTH0808 16 8 8 P01-P08 P09-P16
Total 32 SIP Trunks supported in all Configurations
SETU VFXTH Configurations
Hardware Architecture
Overview
Interfaces
Port Configuration
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
SETU VFXTH1616 Hardware Architecture
Input Supply : DC Power Jack 24 V, 2.5A
Power Supply OP V : +5V +3.3V, -27V -87V
Ethernet Port
FXS Modules: Each Module supports 2 extensions to be connected (Total 8)
FXO Modules (Total 8)
VoIP module : CODEC IC & SDRAM. Total 4 such Modules
Each supporting 8 channels
32 BIT RISC PROCESSOR
FLASH 32MB
128 MB
RAM
CPLD
LED Indications
Overview
Interfaces
Port Configuration
Hardware Architecture
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
Total 34 LEDs in SETU VFXTH1616
Power LED : At Power On Power LED will Turn On
(Continuous Green)
32 Port LEDs : FXO and FXS Port LEDS
At Initialization:
P01-P32 : OFF
After approx 16 sec P01-P03 Glow Continuous Red
After approx 20 sec remaining P04-P32 Glow Red Continuous
After 5 Sec : P01 - P32 LED will be Off
LED Indications
32 FXO/FXS Ports LED indications during normal functioning
Continuously Off Port Idle / Disable
400 ms Red on -
200 ms off -
400 ms RED on -
3000 ms off (2 Blinks)
Incoming Ring Event
400 ms on-
400 ms off (continuous) Red
Off-Hook Event (Dialing State)
Continuous On Red Speech
LED Indications
System
LED (STS)
Installations Do’s and Don’ts
Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
Dust Proof, Moisture Free Location
Away from electromagnetic Sources
Ventilated Location
Path to Static Charges
Stable Mains Supply
Proper Mains Earth
Proper Telecom Earth
Installation DO’s
Installation DONT’s
Applications
Overview
Interfaces
Port Configuration
Hardware Architecture
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
LED Indications
Installation Do’s and Don'ts
Broadband Modem/Router
FXO1…FXO16
PSTN Network
2001
2016
FXS1…FXS16
IP Network
Ethernet
Stand Alone Call Possibilities
1. FXS IP Network
2. FXS PSTN Network
Stand Alone Application
Broadband Modem/Router
FXO1…FXO16
PSTN Network
FXS1…FXS16
IP Network
Ethernet FXS ports of SETU VFXTH are
connected to FXO Ports of PBX. Extensions of PBX thus
can avail the PSTN & VoIP Networks of SETU VFXTH
PBX
FXS1…FXSN
SETU VFXT1616
In Front of PBX
Application
Behind the PBX Application
Broadband Modem/Router
FXO1…FXO16
IP Network
Ethernet FXO ports of SETU VFXTH are connected to FXS
Ports of PBX. Extensions of SETU VFXTH
thus can use the Trunk of SETU VFXTH
PBX
FXS1…FXSN
2001
2016
FXS1…FXS16
PSTN N/W
SETU VFXT1616
Analog Extension PBX Over IP Application
Ethernet PBX FXS1…FXSN
2001
2016
PSTN N/W
SETU VFXT1616
FXO1… FXO16
FXO1… FXO16
Eth
ern
et
2001
FXS1… FXS16
FXO1… FXO16
SETU VFXT1616
IP Network Broadband
Modem/Router
Broadband Modem/Router
Peer to Peer Calling
Ethernet
2001
2001
FXS1… FXS16
FXO1… FXO16
FXS1… FXS16
Ethernet
FXO1… FXO16
2016
IP Network
SETU VFXT1616
SETU VFXT1616
PSTN Call over IP (Long Distance converted to Local Call)
Ethernet
2001
2001
FXS1… FXS16
FXO1… FXO16
FXS1… FXS16
Ethernet
FXO1… FXO16
2016
IP Network
SETU VFXT1616
SETU VFXT1616
Mumbai Delhi
Programming Using Phone
Certain parameters of SETU VFXTH can be configured by dialing system commands
from a telephone connected to the FXS port
You can configure certain network parameters like IP address, Subnet Mask,
Connection Type, set the system to default and also view current IP address, Subnet
Mask, Connection Type, DNS and Gateway address by dialing system commands
Programming Using Phone
SE Login
Connect Analog Phone to FXS port of SETU VFXTH
OFF - Hook the phone
Hear Dial Tone [Toooooooooooooooooooooo]
Dial Command “#19 – SE Password” for Login
Default SE Password is “1234”
Enter System Commands to perform different functions
Dial “00#*” to Exit from Programming mode
Commands
11 – IP Address – #* (To change IP Address)
12 – Subnet Mask – #* (To change Subnet Mask)
10 – Code – #* (to change the connection type) [1 – static, 2 – DHCP, 3 – PPPoE]
31 – Code – #* (To Enable1/Disable0 VLAN Tag)
51 – Reverse SE Password – #* (To Restore Factory defaults)
21 – #* (To view IP Address) & go On – Hook
22 – #* (To view Subnet Mask) & go On – Hook
23 – #* (To view Gateway Address) & go On – Hook
24 – #* (To view DNS Address) & go On – Hook
20 – #* (to view the connection type) & go On – hook
27 – SIP Trunk Number (1 – 9) – #* & go On – hook (To view the status of SIP Trunk)
Commands
programming Using pc
Web Jeeves Login from Local Network
Network Switch
192.168.50.200
192.168.50.33
SETU VFXTH is located on Local IP
Internet 203.88.123.231
SETU VFXTH is located on Global IP
PC with internet connection
Web Jeeves Login from Public Network
Internet
WAN: 203.88.123.231:80
PC with internet connection
IP : 192.168.1.151 Subnet : 255.255.255.0 Gateway : 192.168.1.1
LAN: 192.168.1.1
Router’s port:80 is forwarded to IP Address of
SETU VFXTH
Web Jeeves Login from Public Network
Programming
Built – in Web server
GUI based software called Jeeves
Accessible using any web browser
Default IP of Ethernet Port is 192.168.001.136
Default SE password is 1234
Programming
Enter Ethernet Port IP Address of SETU VFXTH
Programming
Enter Password for Login (Default: 1234)
Login Page
Home Page
Basic Settings
Network Port Parameters
Select Region of system installation and accordingly Call progress tone country wise
• This parameters can be programmed as per existing data network
• Connection type :
1. Static: IP address, Subnet mask & Gateway Address assigned Manually
2. DHCP: IP address, Subnet mask & Gateway Address assigned automatically by
DHCP server
3. PPPoE: Select this option if your ISP provides internet services using PPPoE, If
you select this option you must enter the User ID, password and service name in
PPPoE parameters
Network Port Parameters
Network Port Parameters
Select connection type of SETU VFXTH and according to the connection type program the IP details
Login Password
Password for Jeeves/FTP/Telnet can be minimum of 4 characters and Maximum of 16 characters long
All ASCII characters are allowed except white space & ( ) ; “ ‘ < > | \ dot (.)
Date – Time Settings
Click on arrow to Set date and time manually
Set SNTP server address here to sync date & time with SNTP server
MWI (Message Wait Indication on SIP Trunk
If you have subscribe for MWI on SIP trunk for the voice mail service by your ITSP then Program Message retrieval number provided by ITSP and port number ion which MWI is to be sent
MWI (Message Wait Indication on SIP Trunk
Incoming call management - SIP trunk - FXO Port
• The process of routing calls originated on FXO port and SIP trunks to the
destination port in SETU VFXTH takes place in two steps:
1. Determination of destination number
2. Determination of destination port
Incoming Call Route
Outgoing on which Number
Destination Number determination
FXO Port
Incoming Call Route options on FXO Port
Destination Number Determination on FXO Port
2 different routings defined here 1. Route all IC
calls (with CLI)
2. Route all IC calls (without CLI)
Without Any Destination Number
Define destination port for routing calls
• Without any Destination Number
• To the Fixed Destination Number
• On the basis of Calling Party Number
• After answering the call and collecting the digits
Destination Number Determination on FXO Port
• Incoming call on the FXO port
• All calls received on the FXO port are directly routed to the fixed destination
port, configured for this port, regardless of the destination number
Without Any Destination Number
FXO
022 2631725
SETU VFXTH
Without Any Destination Number
2001
FXS
No destination number will be provided, only Destination port will be applied
• 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
FXO
Fixed Destination Number: 471
SIP
471@matrix- pbx.dynalias.org
0265 2630555
Route To a Fixed Destination Number
SETU VFXTH
• 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
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
• 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
FXO
After Answering the call & collecting the digits
SETU VFXTH
Dial Tone
SIP
471
471@matrix- pbx.dynalias.org
0265 2630555
SIP Trunk
Incoming Call Route options on SIP trunk
Destination Number Determination on SIP Trunk
Destination Port Options on SIP trunk
Destination Port Determination on SIP Trunk
2 different routings defined here 1. Route all IC
calls (with CLI)
2. Route all IC calls (without CLI)
• Without any Destination Number
• To a Fixed Destination Number
• On the basis of Calling Party Number
• To the Called Party Number
Destination Number Determination on SIP Trunk
• Incoming call on the SIP Trunk
• All calls received on the SIP Trunk are directly routed to the fixed destination
port, configured for this port, regardless of the destination number
Without Any Destination Number
SIP
022 2631725
SETU VFXTH
2001
FXS
Without Any Destination Number
No destination number will be provided, only Destination port will be applied
• 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
SIP
Fixed Destination Number: 0265 2630555
FXO
0265 2630555
Route To a Fixed Destination Number
SETU VFXTH
471@matrix- pbx.dynalias.org
• 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
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
• 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
SIP
SETU VFXTH
FXO
0265 2630555
0265 2630555
To the Called Party Number
203.88.142.221
02652630555@ 203.88.142.221
Destination Port Determination
Outgoing By which Trunk
• SETU VFXTH supports different methods of determining the destination port
for the calls originated on FXS Port, FXO Port and SIP trunks, they are:
1. Fixed
2. On the basis of destination number
3. On the basis of calling party number (Not Supported on FXS Port)
Destination Port Determination
Destination Port Determination on FXS Port
Destination Port Options on FXS Port
Destination Port Determination on FXO Port
Destination Port Options on FXO Port
Destination Port Options on SIP trunk
Destination Port Determination on SIP Trunk
Outgoing Call Management - FXS Port - FXO Port - SIP trunk
FXS Port
For OG Call we can allow or block outgoing calls, enable flag to Block the Outgoing from this trunk
FXO Port
Enable flag to Block the Outgoing call from this trunk, Apply ANT with Dialed & substitute number string
SIP Trunk
Options Related to Outgoing calls through SIP trunk
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
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
STUN
Program the STUN Server IP Address & Port here
Select NAT type as STUN if you want to use IP address fetched using STUN
STUN
Status page will display the IP address, port number and NAT type fetched using STUN
STUN
Router Public IP Address
Port Forwarding:
Since STUN doesn’t work with symmetric NAT , as an alternative to STUN Port
Forwarding can be done in the router and Router’s Public address that is configured
can be used as Source Port IP Address
VoIP Port Parameters: Router’s Public IP Address
Router Public IP Address
Use NAT type as Router Public IP address
Router Public IP Address
Program Router Public IP Address here
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 VFXTH IP: 192.168.200.195 G/W : 192.168.200.210
Router separates Private and Public
Network
Private IP
Public IP
203.88.142.218
Port Forward in Router
LAN port of Router WAN 203.88.142.221
*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router Configuration : Example
Router’s Network
Parameters
Port Forwarding:
Router’s SIP and RTP Ports are forwarded to Private IP of SETU VFXTH
*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router Configuration : Example
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
Peer to Peer Calling
Peer to Peer Calling
Program SIP trunk mode as peer to peer for peer to peer calling
Enable SIP trunk
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
Proxy Calling
Requirement for Proxy Calling
Proxy server authenticates the clients for outgoing calls through it
What is required for
authentication?
SIP ID
Authentication ID
Authentication Password
Registrar Server Address
Registrar Server port
Proxy Calling
Select SIP Trunk as Proxy and assign the
authentication credentials provided by service provider
Enable the flag
Proxy Calling
If this flag is enabled, SETU VFXTH will send
the REGISTRAR MESSAGE to
Registrar Proxy as applicable
SIP Registration
On enabling the flag of SIP Registration, following parameters are to be taken care of
This is the time period after which system will send registration request
to maintain binding with Registrar Server. Valid range: 00001-65535.
Default:3600 Seconds
When a registration attempt fails, system resends request to registrar server after this timer’s expiry. Valid
range: 00001-65535. Default:10 Seconds
SIP Registration
System will get unregistered with the current server & will
register with the alternate server, if fallback occurs while sending INVITE message when
Switch Registration to Alternate Server on Fallback is
enabled
Registrar Settings
If you want the system to send
DNS SRV query to the configured domain server, enable this flag
What is DNS SRV?
Dialing by domain names lets a SIP user have a single public “SIP Address” which
can be redirected at will to their current location.
SRV records maintain stability and also opens up the possibility to use your own
domain, regardless of the domain of the SIP service you are using
SIP Registration
Enable the flag, if your service
provider supports multiple servers
in its network
Advance Settings
• Access code is a string of digits dialed to use a feature
• SETU VFXTH users can access the features and facilities by dialing the access
code assigned to them from a phone. User can
1. Enable/Disable a feature
2. Access Supplementary feature
3. Enter into the programming mode
• SETU VFXTH provides default access code for all features, you can change it to
suit your preferences
Access Codes
Access Codes
Access codes can be changed from here
Access Codes
Access codes can be changed from here
• This feature provides the flexibility to allow or deny dialing of a particular
number or a set of numbers from a particular port or all ports
• Allowed Denied number logic makes use of two number lists:
1. Allowed Numbers List: this is the list of numbers that can be dialed out from
the SIP trunk (default number list – 7)
2. Denied Numbers List: this is the list of numbers that are to be restricted from
being dialed out from the SIP trunk (default number list – 8)
Allowed – Denied Numbers
Allowed – Denied Logic on FXS Port
Apply allowed denied list on FXS Port & program the number list for allowed & denied numbers
Apply allowed denied list on FXO port & program the number list for allowed & denied numbers
Allowed – Denied Logic on FXO Port
Allowed – Denied Logic on SIP Trunk
Apply allowed denied list on SIP trunk & program the number list for allowed & denied numbers
• This feature is used to translate the dialed number string to preprogrammed
number string
• ANT can be used to modify, add or delete the prefix of the destination number
string
• For this feature we need to configure dialed number string and substitute
number string in number list table
• ANT feature is applied on destination ports (On all SIP trunks and FXO Ports)
Automatic Number Translation
Apply ANT on FXO port and program the table number
Automatic Number Translation
Automatic Number Translation
Apply ANT on SIP Trunk and program the table number
ANT table
Automatic Number Translation
Examples on how to program
• SETU VFXTH supports feature ‘Black listed Callers’ which enables you to block
incoming calls from specific numbers and addresses on the SIP trunks
• This feature is applicable on source port only
• To use this feature, user must configure the numbers of unwanted callers in a
number list
• Enable the Reject Calls from Blacklisted Caller check box on the SIP trunks on
which you want to apply this feature
Black Listed Callers
Black Listed Callers
Apply black listed caller feature on selected SIP trunk and define the number list for the same Black Listed Callers
• It’s a record for the calls, containing information about the gateway’s usage
when call was made
• Maximum of 2000 call record entries can be stored
• Call record entries are stored in FIFO logic
• User can set different filters as required and generate Call Detail Record (CDR)
report
• Call records can be cleared manually at any time
Call Detail Record (CDR)
• It is possible to get following details of a call with CDR
1. Date of call origination
2. Time of call origination
3. Calling number
4. Called number
5. Duration of call
6. Source port
7. Destination port
8. Disconnected by
9. Cause
10. PIN number
11. Remarks
Call Detail Record (CDR)
• 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)
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
CDR can also be viewed from Jeeves
Call Detail Record (CDR)
• PIN authentication is a security feature to restrict access to the system and
prevent possible misuse of resources
• User can use the PIN authentication on the source port to establish identity of
callers before their call is processed by SETU VFXTH
• PIN authentication can be used on the source port only if the incoming call
routing for the source port is set to After answering the call and collecting digits
• To use this feature it must be enabled on the source port and the PIN
authentication table must be configured
PIN Authentication
• The PIN authentication table stores up to 500 PIN numbers and their
corresponding authentication passwords
• If PIN authentication is enabled on source port, SETU VFXTH answers the
Incoming call and plays a feature tone, it waits for the caller to dial the PIN
number and password, it matches them with the PIN authentication table, if
match is found it allows the call to be processed
• In case of wrong PIN entered, SETU VFXTH allows the caller to make two more
attempts, if the caller fails to dial correct PIN and password in all attempts, the
system disconnects the call
PIN Authentication
PIN Authentication – FXO Port
Select routing type ‘after answering the call and collecting the digits’ for PIN authentication feature to use
Enable this flag for prompting caller to enter PIN
PIN Authentication
Enter PIN number & PIN password, system checks PIN entered by the caller during call with the entries in the PIN authentication table, if match found then only the call will be processed further
• Digest authentication is a challenge – based authentication service of SIP to
authenticate the identity of the originator of SIP request in the INVITE message
• The recipient of the request can ascertain whether or not the originator of the
request is authorized to make the request
• When the digest credentials of the originator – User Name and Password – in
the INVITE message are authenticated and accepted by the recipient, the
originator and recipient are connected
• You may use the digest authentication to restrict access to SETU VFXTH to
specific callers, prevent unwanted or malicious calls
Digest Authentication
• When this feature is enabled on a SIP trunk for all Incoming calls
1. SETU VFXTH will challenge the identity of the calling party
2. When the calling party sends its credentials, SETU VFXTH authenticates the
credentials by matching it with its Digest Authentication table
3. If a match is found, the calling party will be authenticated and the call will be
allowed on the SIP trunk
4. If no match is found, SETU VFXTH will consider it as invalid authentication
information and reject the call
Digest Authentication
Enable apply flag in SIP trunk to use digest authentication
Digest Authentication
Enter Digest credentials (User ID and User Password) of calling party
Digest Authentication
• Static Routing Table is required when you have more than one router (Gateway)
in your network and you want SETU VFXTH to send packets to multiple
routers/gateways for different types of calls
• If you have only one router connected in the network , you need not configure
this table & LAN interface of router will act as the default gateway for the system
Static Routing
Program the static routing table with the details, if the match is found here then gateway will send the packets to defined gateway address opposite to the destination address
Static Routing
• Prefix to domain name conversion is used when a user sets call forward or
makes a blind transfer on SIP, this feature is applicable only when the
destination port is SIP
• SETU VFXTH supports multiple SIP trunks & FXS ports, when a FXS port user dials
a SIP number, SETU VFXTH routes the call to the IP network using the SIP trunk
determined by the routing mechanism. The FXS user can dial only numbers not
domain names, therefore it becomes necessary that the domain names be
assigned prefix codes which the FXS user can dial
Prefix to Domain Name Conversion
• User need to program prefix v/s domain name in the table
• This table is not checked for making an outgoing call, but it is checked when
some FXS port has set call forward and only number is programmed or user is
doing blind transfer
• For example prefix in the table is programmed as *123 and domain name as
abc.com and destination number for call forward is *1239974 then it will be
replaced by 9974@abc.com
Prefix to Domain Name Conversion
Define prefix and domain name in the table
Prefix to Domain Name Conversion
• If call disconnection is signaled by your CO network in the form of disconnect
tone on the FXO Ports
• You must enable Disconnect Tone Detection on the FXO port and select the
Disconnect tone type
• To enable the system to detect the disconnect tone accurately, you must
configure the cadence and frequency of the disconnect tone type you selected,
as supported by the CO network
Disconnect Tone
Enable disconnection tone detection here
Disconnect Tone
Disconnect Tone
Program the disconnect tone cadence here
• SETU VFXTH supports dialing of emergency numbers from all ports, Emergency
numbers and their respective routing groups must be configured in the
emergency number table
• User can configure up to 10 numbers of emergency services such as ambulance,
fire brigade, police etc.
• By default, No emergency numbers are loaded in the system, in the emergency
number table
Emergency Numbers
Click here to add new entry to the table Click here to delete
entry from the table
Click here to Edit entry of the table
Emergency Numbers
Features
• If any FXS port want to use supplementary services then these services must
be activated in COS for particular FXS port as well as at SIP services provider in
case of SIP account calling
• SETU VFXTH offers following telephony features, which they can access by
dialing access codes
1. Call Hold 6. Blind Transfer
2. Call Forward 7. Attended Transfer
3. Call toggle 8. Do Not Disturb (DND)
4. Call waiting 9. Hotline
5. Conference
Class Of Service
Enable required feature from Class of service on particular FXS port
Class Of Service
Enable the supplementary services after enabling the feature in COS
Supplementary Services
• When SETU VFXTH is interfaced with service provider server – ITSP or other
PBX that supports supplementary services that require dialing of Flash like call
hold, call transfer, call waiting, you must select the subscriber type according to
the extent of feature access you want on the FXS port connected to the system
Subscriber Type
• Select Network if you want to use supplementary services supported by the
other PBX, you can access the service provider features by dialing FLASH, you
will not be able to access the local features of SETU VFXTH
• Select Gateway if you want to use supplementary services supported by the
SETU VFXTH, in the gateway mode you will also be able to access the
supplementary services of the service provider which require dialing of FLASH
Subscriber Type
Select the subscriber type of your choice
Subscriber Type
Signaling Loop Start
Connector RJ-45
Off-Hook Line Impedance 600 Ω / 900 Ω / Complex
No. of Long Loop Extension 4
Loop Limit 1800 (Max) Excluding Telephone Set
On-Hook Voltage (Tip/Ring) -48 V
Off-Hook Current 25 mA (Max)
Ringing Voltage Trapezoidal 60 VRMS/25Hz and
Sinusoidal 52VRMS/25Hz
FXS Port
REN 3
DTMF Detection ITU-T Q.24
CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore
Protection Over Voltage Secondary Protection
Return Loss >18 dB
Longitudinal Balance >50 dB
Transmission Level Adjust Tx Gain : -3dB to +6dB; Rx Gain : -3dB to +6dB
Answer Signaling on FXS Battery Reversal
Disconnect Signaling on FXS Battery Reversal & Open Loop Disconnect
FXS Port
Hardware settings on FXS port
FXS Port
General settings on FXS port
FXS Port
First Digit & Inter Digit wait timer
FXS Port
First Digit Wait Timer:
• Signifies the time for which the system waits for receiving a first digit after
going off – hook from FXS port
• On expiry of this timer, system will give error tone to the user
• It is programmable from 01 to 99 seconds (Default: 15 seconds)
FXS
Inter Digit Wait Timer:
• Signifies the time period between 2 consecutive digits while the system is
receiving the digits from caller
• On expiry of this timer, ATA1S will process the digits dialed so far by the user
• it is programmable from 01 to 99 seconds (Default: 5 seconds)
FXS
Return Loss >18 dB
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
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
Hardware settings on FXO port
FXO Port
FXO Port
General settings on FXO port
• This feature enables callers to disconnect the current call and make a new call
using SETU VFXTH without getting disconnected from the system
• This feature is useful when you want to make multiple calls without getting
disconnected each time their call ends
• This feature is applicable only on the FXO port and only when After answering
the call and collecting digits is selected as the destination number
determination method
• If you have enabled Connect source port when number is out dialed on the FXO
port, you will not be able to provide this feature to callers
Making a new call using access code
• To make a new call using access code
In speech with the current call
Dial #91
Current call will disconnect
Dial the new number you want to call
Speech will be establish on the new call as called party answers the call
While in speech dial #91 again to make another new call
Making a new call using access code
Enable the flag to allow user making new call using access code
Making a new call using access code
• SETU VFXTH enables user to disconnect a call using an access code
• When the call disconnect access code is dialed, SETU VFXTH releases the port
engaged in the call
• This feature is applicable only when destination number determination method
is selected as After answering the call and collecting digits
• If you have enabled Connect source port when number is out dialed on the
FXO port or have enabled Connect source port when 183 is received on SIP on
the SIP trunk, you will not be able to provide this feature to users
Disconnecting a call using access code
Disconnecting a call using access code
Enable the flag to allow call disconnection using access code
Enable the flag to allow call disconnection using access code
Disconnecting a call using access code
• SETU VFXTH supports direct dialing of IP addresses from the source port. To
provide IP dialing facility to the users, you must configure a SIP trunk or a SIP
group for IP dialing
• IP number can be dialed with dot ’.’ as entered by ‘*’ while dialing it
• For e.g. to dial IP address 192.167.100.1 dial as 192*167*100*1 from the
Phone at FXS
• When an IP address is dialed from the source port of SETU VFXTH, the system
does not check the destination port determination method you have
configured for that port, instead it routes the dialed IP address through the SIP
trunk or SIP group you configured for IP dialing
IP Dialing
SIP trunk or SIP trunk group can be defined for IP dialing
IP Dialing
100rel and SIP PRACK
SIP PRACK (SIP Provisional Acknowledgement) is a method to enable reliability for SIP 1XX messages
The Called Party answers the PRACK by 200 OK and PRACK is only for 1XX
messages other than 100 Trying
Generally PRACK message flows from Calling Party to Called Party
System Parameters
Enable Provisional acknowledgement for all 1xx messages other then 100 trying
SIP Timers
• SIP Invite Timer
• SIP Provisional Timer
• General Request Timer
SIP Invite Timer
• It is the time for which SETU VFXTH waits for a response from the called party
after sending INVITE message
• This time starts after sending INVITE message to the called party and stops on
receipt of provisional response or final response or when the user goes ON-
Hook, on expiry of the timer the call is disconnected
• The range of SIP INVITE Timer is 10 - 80 seconds (Default: 30 Seconds)
SIP Provisional Timer
• It is the time for which SETU VFXTH waits for final response after receiving
provisional response from the called party
• This timer starts on receipt of provisional response from the called party and
stops on receipt of final response from the called party or when the user goes
ON-Hook, on expiry of the timer the call is disconnected
• The range of SIP Provisional Timer is 10 - 180 seconds (Default: 60 Seconds)
• It is the time for which SETU VFXTH waits for the response of a transaction
request
• This timer starts on initiating a transaction
• This timer stops on receipt of a response for the request
• On expiry of timer, the SETU VFXTH clears the transaction
• The range of SIP Provisional Timer is 10 - 60 seconds (Default: 20 Seconds)
General Request Timer
System Parameters
Program the timer values according to the requirement
SIP Over TCP
• The SIP over TCP option allows you to send/receive the SIP messages over TCP
• SIP over TCP is applicable for both Proxy and Peer to Peer
• By Default SIP messages transported over TCP
• Disable the flag to send SIP messages over UDP
SIP Over TLS
• The SIP over TCP option allows you to send/receive the SIP messages over TLS.
TLS protects SIP signaling against loss of integrity, confidentiality and against reply
• SIP over TLS is applicable for both Proxy and Peer to Peer
• By Default SIP over TLS is enabled
• Disable the flag to disable SIP over TLS
System Parameters
Program SIP TCP, UDP and TLS port values. Default: 5060 for TCP, UDP and 5061 for TLS
Server Port
Server ports can be changed to any value from 1021 to 65,535
Management/Security
Server ports can be changed to any value from 1021 to 65,535
Certificate
Certificate
• SETU VFXTH supports certification for TLS, Web Server, Firmware Upgrade,
Configuration Upgrade and TR-069.
• SETU VFXTH supports two types of Certificates: Self-Signed Certificate and CA
Signed Certificate.
Self – Signed Certificate
• A self-signed certificate is created by the clients themselves or by the Servers and
then given to their clients.
• It means that you yourself become the Certificate Authority (CA), create a CA
Certificate and sign it.
• The self-signed certificate is faster to create but is not signed by a trusted CA
Organization.
• The self-signed certificate must be installed in the trusted list of clients that
connects over TLS with the Server. Because the certificate has been self signed, the
signature is not likely to be in the clients’ trust file, hence, they need to add it.
Certificate
Generate self signed CA certificate by entering the required details below
Once you generate self-signed certificate, you must send it to your clients so that they install it in their trusted list.
Click generate to generate new certificate for entered details
Certificate
System will show generated certificates under trusted root CA
System Certificate
• After creating a Self-Signed CA Certificate, you can either,
• Generate a System Certificate for your clients. These System Certificates can then be given to the respective clients OR
• The Clients can prepare their own System Certificates. For this you need to send them the CA Certificate created by you OR
• Generate a Certificate Signing Request (CSR), if you want the Certificate to be signed by a third party
If the clients prepare their own certificates, you need to send your CA Certificate to all the clients. The clients must upload the same in their system. Similarly, all the clients must send their CA Certificates to you and you must upload the same in your system. To avoid this, it is recommended that you create the Certificates and then provide it to your clients
Enter details to generate system certificate
If you want to get a CA Signed Certificate, you need to do the following: 1. Generate and enroll the Certificate Signing Request (CSR). 2. Get the Certificate Signing Request (CSR) verified and signed by the Certified Authority (CA).
Certificate
List of available system certificates
User can also upload the certificates
Certificate
Define the certificate to be used for desired application
Maintenance
Firmware
Browse the ZIP file having new firmware files & click on Upgrade button to upgrade the system firmware
Program the details for Auto firmware upgrade
Upgrade firmware automatically from Matrix Server
Configuration
Browse the ZIP file having configuration files & click on Upgrade button to upgrade the system configuration
Program the details for Auto configuration
Click on Backup Configuration to save config.zip file
• Debugs are logs of actions and events that take place on system, these logs are
useful for troubleshooting and system security
• SETU VFXTH supports Syslog client for debugging, Syslog client enables the
system to send debug messages in Syslog format to the remote ‘Syslog server’
on the IP network
• Syslog uses the UDP as transport protocol
• To be able to use this feature, you must enable ‘Syslog’, configure the Syslog
Server Address and define the server port on which the Syslog will listen for
debug messages
System Debug
System Debug
Debug events can be viewed on the screen
Click debug settings to set parameters for debug and to start debug in PC/Laptop connected to SETU VFXTH
System Debug
Program the IP address and port number of PC/Laptop where Syslog server is installed
Debug for Port: clear the check box to disable the debug for the port which is not needed
• SNMP – Simple Network Management Protocol
• SNMP protocols supported – SNMPV1, SNMPV2C, SNMPV3
• SETU VFXTH is having built in SNMP Server (SNMP Server). It receives SNMP
requests and generates SNMP responses or notifications
• SNMP Manager usually network management station. It generates SNMP
requests and receives SNMP responses and notifications. The SNMP manager is
an SNMP client
SNMP
SNMP
Program SNMP details
System Port Activity
System port activity like Idle, Inactive, Disable, Dial, Speech, ringing, Incoming Call Proceeding, Remote Held, Error
• PCAP or Packet capture consists of intercepting and logging the traffic passing
over the network, PCAP intercepts each packet in the data streams that flow
across the network, and can decode and analyze its contents
• A maximum 2MB of packets can be captured and stored in the system
• SETU VFXTH also supports filters and promiscuous mode for capturing packets
• If promiscuous mode is enabled, SETU VFXTH will capture all network traffic and
if disabled then system will capture only traffic that is directly related to SETU
VFXTH (to or from SETU VFXTH)
PCAP Trace
PCAP Trace
Click here to start the PCAP trace Click here to stop
the PCAP trace
Once the PCAP is captured save the trace file on your PC/Laptop
Click here to Enable Promiscuous mode
Enter the filter details here
• Select source port and destination port with source number and destination
number.
• When Call button from GUI is pressed system will call source number first and
when answered by source port it will ring on destination port & speech path
can be checked
• Clicking on call button will also lead the programmer to system port activity
page to monitor the status of the port during call progress
Manual Call Test
Manual Call Test
• SETU VFXTH supports the AC Impedance Test for clear, audible and echo-free
speech over FXO Ports.
• This test helps you to set the most appropriate values for the FXO Port
Parameters —AC Termination Impedance, CO Termination and CO Line Type— to
correct the line impedance mismatch between the AC Termination Impedance
presented by the FXO Port of SETU VFXTH to the line and the CO Termination
Impedance presented by the Central Office to the line.
• While the test is being conducted, you will hear pulsating tone on all the ports of
the system. (Mute the microphone of destination landline number or mobile
number when call is answered by destination number)
AC Impedance Test (FXO)
AC Impedance Test (FXO) Enter phone number on which system will make the call in order to complete the test
Select the FXO port on which you want to run the test
Click on start test and wait for the results
For more details click help
Default System
Click OK to factory default the gateway
Soft Restart
Click OK to Restart SETU VFXTH
• TR-069, also known as CPE WAN Management Protocol (CWMP), is a remote
management protocol used for secure communication between the Customer
Premises Equipment (CPE) and an Auto-Configuration Server (ACS) for various
functionalities such as:
Auto-configuration and dynamic service provisioning
Firmware Management
Status and performance monitoring
Diagnostics
• SETU VFXTH supports TR-069. Using TR-069, service providers can manage SETU
VFXTH remotely for the functions described above.
TR – 069
TR – 069
Program TR-069 details
STATUS
System Detail
Version Revision details
Firmware Status
Last Firmware up gradation details if scheduled firmware upgrade is ON
Configuration Status
Last Configuration up gradation details if scheduled firmware upgrade is ON
Network Status
IP details status of IP configured in SETU VFXTH
FXO Port Status
Line connection status on FXO
SIP Trunk Status
SIP trunk Status
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