Matrix SETU ATA211 V1 System Manual 111
'Route to fixed destination number through FXO'
• This feature is required when user changes his location and wants to receive all IC calls to the new
location on PSTN/Mobile network (where IP connectivity is not available at new location).
• If this option is selected, the user should program the desired number in field for 'Fixed Destination
number' where he wants to receive the calls. On receipt of call on SIP, ATA will immediately place the call
on the programmed fixed destination number through the FXO Port. (If FXO is busy or disable, call will be
rejected).
'Route to FXS. If not answered within ring timer, route to fixed destination number through FXO'
• This feature is a replica of Call Forward-No Reply with a difference that user can route only calls to PSTN
network (call is routed to different network).This feature is generally required when user frequently
changes his original location, where internet is not accessible.
• If this option is set, the IC call will be first routed to FXS Port. If it is not answered with in ring timer, call will
be automatically routed to the fixed destination number through the FXO Port. For this option, user should
program the desired PSTN/Mobile number as the destination number. (If FXO is busy or disable, call will
be rejected).
'Answer the call and connect to FXO'
• This option is required when user wants to make PSTN call from the remote location where only IP
connectivity is available.
• If this option is set, on receipt of the IC call on SIP account, the caller will immediately get connected to
FXO port (PSTN network). The caller will get dial tone of PSTN (if FXO port of ATA211 is connected to
PSTN exchange) OR caller will get dial tone of PBX (If FXO port of ATA211 is connected to FXS port of
PBX).
• Now, the caller can dial the desired contact number. (If FXO port is busy or disable, call will be rejected).
'Route to FXS. If not answered within ring timer, answer the call and connect to FXO'
• This option is desirable in the field applications when user wants to make PSTN call from the remote
location.
• If this option is set, on receipt of the IC call on SIP account, the caller will get RBT for the programmed ring
timer and if call is not attended during this time, he will get connected to FXO port (PSTN network). The
caller will get dial tone of PSTN (if FXO port of ATA211 is connected to PSTN exchange) OR caller will get
dial tone of PBX (if FXO port of ATA211 is connected to FXS port of PBX).
• Now, the caller can dial the desired contact number. (If FXO is busy or disable, call will be rejected).
'Collect number and route to FXS'
• This routing type will be required when Authentication Type configured is 'PIN' type and PIN authentication
is to be used on SIP for the calls on FXS from SIP.
• If this option is set, the caller will get dial tone on receipt of the IC call on SIP account. For this option, while
processing the call the ATA211 will collect the number string from the caller and then route the call to FXS
port.
'Collect number and route to FXO'
• This routing will be required when Authentication Type configured is 'PIN' type and PIN authentication is to
be used for SIP for the calls on FXO from SIP.
• If this option is set, the caller can dial SIP Number of ATA211 and make PSTN call from the remote
location. For this option, while processing the call the ATA211 will play dial tone, collect the number string
from the caller and then route the call to FXO port.
'Route as per called party number'
• This routing type will be required when FXS port of ATA211 is connected to the one of the FXO port of PBX
or when FXO port of ATA211 is connected to PSTN OR to the FXS port of PBX.
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