System information

Advanced Options for Voice Services
VoIP-to-PSTN and PSTN-to-VoIP Calling
Cisco SPA232D Administration Guide 209
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PSTN Call to Ring Line 1
This feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the
PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops. After a given
number of rings, the VoIP gateway picks up the call.
Symmetric RTP
The Symmetric RTP parameter is used to send audio RTP to the source IP and
port of the inbound RTP packets. This facilitates NAT traversal. You can configure
these settings in the SIP Parameters section of the SIP page.
Call Progress Tones
The ATA has configurable call progress tones. Call progress tones are generated
locally on the ATA, so an end user is advised of status (such as ringback).
Parameters for each type of tone (for instance a dial tone played back to an end
user) may include the following specifications:
• Number of frequency components
• Frequency and amplitude of each component
• Cadence information
When one VoIP account is shared between the FXS and PSTN Lines, the following
parameters are recommended to be set. You can configure these settings on the
Regional page.
Call Progress Tone Description
VoIP PIN Tone This tone is played to prompt a VoIP caller to enter a PIN number.
PSTN PIN Tone This tone is played to prompt a PSTN caller to enter a PIN
number.
Outside Dial Tone During two-stage PSTN-gateway dialing and with a dial plan
assigned, the ATA device collects digits from the VoIP caller and
processes the number using the dial plan. The ATA device plays
the
Outside Dial Tone to prompt the VoIP caller to enter the
PSTN number. This tone should be specified to sound different
from the PSTN dial tone.