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TA1610 User Manual
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route dial rules.
Calls from PSTN to SIP
1) Create an inbound route on your SIP server, and select the SIP trunk you
have registered just now.
2) Configure a
“Port->IP” route on TA1610, choose a PSTN trunk or PSTN trunk
group
in the field “Call Source”, and choose the SIP account in the filed “Call
Destination
”.
3) When a call comes to PSTN trunk on TA1610, the call will be routed to the
destination of the SIP server inbound route.
Register SIP account on IP phone
With account mode, you can directly take the SIP account to register on your SIP
phone or softphone; then make calls from softphone though PSTN trunk on
TA1610 and receive incoming calls on your SIP phone or softphone. In this way,
you don’t have to set up any SIP server.
VoIP Mode
Take a SIP account from your SIP server, and register it on TA1610 as a VoIP trunk. In
this way, TA1610 and your SIP server are connected.
Calls from SIP to PSTN
1) Configure a IP-> Port route on TA1610; choose the VoIP trunk
in the field “Call
Source
”, and choose PSTN trunk in the field “Call Destination”.
Enable
Two-stage Dialing
on the route.
2) Make a call from your SIP server, dial the SIP account number which is
registered on TA1610. You will hear a dial tone, then dial the external number
out through PSTN trunk.
Calls from PSTN to SIP
1) Configure a Port->IP route on TA1610, choose PSTN trunk
in the field “Call
Source
”, and choose the SIP trunk in the field “Call Destination”.
2) When an incoming call reaches PSTN trunk on TA1610, you will hear a dial
tone, then dial an extension number of the SIP server.
SPS Mode
(Recommended)
Create a Service Provider SIP trunk on TA1610 to connect to your SIP server. Add
another Service Provider SIP trunk on your SIP server, connecting to TA1610.
Calls from SIP to PSTN
1) Create one outbound route on your SIP sever, and select the SIP trunk you
have created just now.
2) Configure a IP->Port route on TA1610
, choose the SPS trunk in the field “Call
Source
”, and choose PSTN trunk in the field “Call Destination”.
3) Make a call from your SIP Server and the call should match the outbound
route dial rules.
Calls from PSTN to SIP