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Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
ATA Administration Guide 95
6
• VoIP user—VoIP caller that has a user account (user-id and password) on the
ATA de vice
• PSTN caller—One who calls the ATA device from the PSTN to obtain VoIP
service
Line 1 can be configured with a regular VoIP account and can be used in the same
way as the Line 1 of any ATA device.
A second VoIP account can be configured to support PSTN gateway calls
exclusively. A different SIP port should be assigned to Line 1 and the PSTN Line.
The same VoIP account may be used for both Line 1 and the PSTN Line if a
different SIP port is assigned to each.
VoIP callers can be authenticated by one of the following methods:
• No Authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
• HTTP digest—SIP INVITE must contain a valid authorization header.
PSTN callers can be authenticated by one of the following methods:
• No authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the SPA3102, the VoIP caller establishes a
connection with the PSTN Line by way of a standard SIP INVITE request
addressed to the PSTN Line. The PSTN Line can be configured to support one-
stage and two-stage dialing as described in the following sections.
One-Stage Dialing
One-stage dialing allows a call to be started over VoIP and then immediately get a
dial tone on the PSTN.
To use one-stage dialing, the Request-URI of the INVITE to the PSTN Line should
have the form <
Dialed-Number
>@<
SPA-Address
>, where <
Dialed-Number
> is the
number dialed by the VoIP caller, and <
SPA-Address
> is a valid address and port
of the SPA3102, such as 10.0.0.100:5061.