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Advanced Options for Voice Services
VoIP-to-PSTN and PSTN-to-VoIP Calling
Cisco SPA232D Administration Guide 217
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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.
NOTE When the source address of the INVITE is 127.0.0.1, authentication is automatically
disabled because this is a call by the local user. This applies to both one-stage and
two-stage dialing.
You can configure these settings in the VoIP-To-PSTN Gateway Setup section of
the PSTN (LINE Port) page.
How PSTN-To-VoIP Calls Work
For PSTN-to-VoIP calls, the basic PSTN-to-VoIP call flow is as follows:
1. When a PSTN call comes in to the ATA device and is unanswered (after a
configurable number of rings), then the ATA takes the LINE port off hook.
2. The ATA device plays dial tone.
3. The PSTN caller enters the target telephone number. The collected digits are
processed by the default dial plan.
You can add PIN authentication to the basic flow:
1. When a PSTN call comes in to the ATA and is unanswered (after a configurable
number of rings), then the ATA takes the LINE port off hook.
2. The ATA prompts the caller to enter the PIN number followed by the # key.
3. The ATA compares the PIN to the configured PSTN PIN values.
• If the PIN matches one of the configured PSTN PIN values, then the ATA
plays dial tone. The caller enters the telephone number and the collected
digits are processed by the dial plan associated with the PIN number.
(These dial plans are configured on the Voice > PSTN page, Dial Plans
section.)