Specifications

28 Matrix SETU ATA211 System Manual
• Route to fixed destination number through FXO: The system will place all calls received on the SIP
Trunk to a Fixed Destination Number through the FXO port.
If you select this option, define the destination number in the Fixed Destination Number field.
• Route to FXS. If not answered within ring timer, route to fixed destination number through FXO:
The system will place all calls received on the SIP Trunk on the FXS port. The phone connected to the
port will start ringing for the duration of the Ring Timer (default: 45 seconds). If the call is not answered
within this timer, the call will be automatically routed to the fixed destination number through the FXO
port.
If you select this option, define the destination number in the Fixed Destination Number field.
• Answer the call and connect to FXO: The system will connect all calls received on the SIP Trunk on
the FXO port. You will get dial tone of the PSTN, if the FXO port is connected to the PSTN
exchange.You can now dial the desired number.
• Route to FXS. If not answered within ring timer, answer the call and connect to FXO: The system
will place all calls received on the SIP Trunk on the FXS port. The phone connected to the port will start
ringing for the duration of the Ring Timer (default: 45 seconds). If the call is not answered within this
timer, the system will automatically connect the call to the FXO port. You will get dial tone of the PSTN
if the FXO port is connected to PSTN exchange. You can now dial the desired number.
• Answer the call, collect number & route to FXS: Select this option, if FXS port of SETU ATA211 is
connected to the FXO port of the PBX and you want callers to reach extensions of the PBX by simply
dialing the extension number. When there is an incoming call on the SIP trunk, the call is answered and
dial tone is played to the caller. When the caller dials the number, ATA will collect the digits and route
the call from its FXS port.
• Answer the call, collect number & route to FXO: Select this option, if you want the system to first
answer the call on the SIP trunk, play dial tone to the caller, and when the caller dials the number, to
collect the digits and route the call to the FXO port.
• Route as per called party number: Select this option, if you want all incoming calls on the SIP trunk to
be routed on the basis of the Called Party Number received in INVITE message.
If you select this option, you must configure the Called Party Number Table.
To do this,
• Click the arrow icon. The Called Party Number Table opens in a new window.
• In this table, entries are stored at Index numbers from 02 to 10.
• For each index number,
• enter the desired Number (max. 24 characters) you want to store in the table.
The first Index number 01, is reserved for No Match Found. Start your entries from Index 02.
• For the number you entered, define the Minimum Digits, which the system must receive to
consider it as a valid number. Default:01