Cisco SPA2102-SF Administration Guide - Page 95

Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work 6 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. These settings can be configured on the SPA3102 PSTN Line page. See "VoIPTo-PSTN Gateway Setup section," on page 210. 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 device takes the FXO 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. On the SPA3102, you can add PIN authentication to the basic flow: 1. When a PSTN call comes in to the ATA device and is unanswered (after a configurable number of rings), then the ATA device takes the FXO port off hook. 2. The SPA3102 prompts the caller to enter the PIN number followed by the # key. 3. The SPA3102 compares the PIN to the configured PSTN PIN values. • If the PIN matches one of the configured PSTN PIN values, then the SPA3102 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 Voice tab > PSTN Line page, Dial Plans section.) • If the PIN does not match one of the configured PSTN PIN values, then the ATA device plays the reorder tone and then takes the FXO port on-hook. Cisco Small Business ATA Administration Guide 95