Chapter 4 Configuring The Pstn Gateway (Fxo); How Voip-To-Pstn Calls Work; One-Stage Dialing; Document Version - Linksys SPA1001 Administrator User Manual

How VoIP-To-PSTN Calls Work

Line 1 can be configured with a regular VoIP account and can be used in the same way as the Line 1 of
any Linksys ATA.
With the SPA3102 and AG310, 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:
•
•
•
PSTN callers can be authenticated by one of the following methods:
•
•
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

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 of the SPA3102, such as 10.0.0.100:5061.
If the FXO port is currently in use (off-hook) or the PSTN line is being used by another extension, the
SPA3102 replies to the INVITE with a 503 response. Otherwise, it compares the <Dialed-Number> with
the <User ID> of the PSTN Line. If they are the same, the SPA3102 interprets this as a request for
two-stage dialing (see the
processes the <Dialed-Number> using the corresponding <Dial Plan>.
If dial plan processing fails, the SPA3102 replies with a 403 response. Otherwise, it replies with a 200
and at the same time takes the FXO port off hook and dials the target number returned after processing
the dial plan.
If <User ID> on the PSTN Line is blank, <Registration> should be disabled for the PSTN Line.
Note
If HTTP Digest Authentication is enabled, the SPA3102 challenges the INVITE with a 401 response if
it does not have a valid Authorization header. The Authorization header should include a <User ID n>
parameter, where n refers to one of eight VoIP user accounts that can be configured on the SPA3102.
The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The
<User ID n> must match one of the VoIP accounts stored on the SPA3102. Each VoIP user account
contains the information listed in
Linksys ATA Administrator Guide
4-2
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
No authentication—All callers are accepted for service
PIN—Caller is prompted to enter a PIN right after the call is answered
"Two-Stage Dialing" section on page
Table 1-27.
Chapter 4 Configuring the PSTN Gateway (FXO)
4-3). If they are different, the SPA3102

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