System information

Advanced Options for Voice Services
VoIP-to-PSTN and PSTN-to-VoIP Calling
Cisco SPA232D Administration Guide 203
C
VoIP-to-PSTN and PSTN-to-VoIP Calling
The ATA allows calls to be made by using SIP-based Voice-over-IP (VoIP) services
and traditional telephone Public Switched Telephone Network (PSTN) services.
Calls can be placed and received by using an analog phone or fax machine and
Cisco SPA302D Mobility Enhanced Cordless Telephone Handsets.
The ATA maintains the state of each call and makes the proper reaction to user
input events (such as on/off hook or hook flash). Because the ATA uses the Session
Initiation Protocol (SIP), it is compatible with most Internet Telephony Service
Provider (ITSP) offerings.
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the Cisco SPA302D, the VoIP caller establishes a
connection with the PSTN Line by way of a standard SIP INVITE request
addressed to the PSTN Line.
One-Stage Dialing
One-stage dialing allows a call to be started over VoIP and then immediately get a
dial tone on the PSTN. When you take a phone off hook and dial a number, the call
is automatically routed to the VoIP or the PSTN, based on the dial plan.
Optionally, you can enable HTTP Digest Authentication. In this case, the ATA
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 ATA device.
Services
Ready
Platform
Telephone/fax
Ethernet
V
V
V
Phone
V
V
PSTN
Internet
Service Provider
VoIP Infrastructure
SIP proxy
IP
Voice
gateway
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