System information

Advanced Options for Voice Services
VoIP-to-PSTN and PSTN-to-VoIP Calling
Cisco SPA232D Administration Guide 204
C
The credentials are computed based on the corresponding password using
Message Digest 5 (MD5). The <
User ID n> parameter must match one of the VoIP
accounts stored on the ATA device. You can configure these settings on the PSTN
(LINE Port) page.
Two-Stage Dialing
In two-stage dialing, the LINE port goes off-hook but does not automatically dial
any digits after accepting the call. To invoke two-stage dialing, the VoIP caller
should INVITE the PSTN line without the user-id in the Request-URI or with a user-
id that matches exactly the <
User ID n> of the PSTN Line. A different user-id in the
Request-URI is treated as a request for one-stage dialing if one-stage dialing is
enabled, or dropped by the ATA (as if no user-id is given) if one-stage dialing is
disabled.
HTTP Digest Authentication can be also used for two-stage dialing, as in one-
stage dialing. If using HTTP Digest Authentication or Authentication is disabled, the
VoIP caller should hear the PSTN dial tone right after the call is answered (by a SIP
200 response).
You also can enable PIN authentication. In this case, the VoIP caller is prompted to
enter a PIN number after the ATA answers the call. The PIN number must end with
a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight
VoIP caller PIN numbers can be configured on the ATA. A dial plan can be selected
for each PIN number. If the caller enters a wrong PIN or the ATA times out waiting
for more PIN digits, the ATA tears down the call immediately with a BYE request.
The call scenarios may involve the following types of callers:
• VoIP caller—Someone who calls the ATA device via VoIP to obtain PSTN
service
• VoIP user—A VoIP caller that has a user account (user-id and password) on
the ATA
• PSTN caller—Someone who calls the ATA device from the PSTN to obtain
VoIP service
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.