Cisco SPA232D Instruction Manual

Cisco SPA232D Mobility Enhanced Phone Adapter
ADMINISTRATION
GUIDE
390487

Cisco SPA232D Administrator Guide 1
Contents
Chapter 1: Getting Started 5
Feature Overview 5
Understanding Voice Service Operations 6
ATA Voice Features 6
Before You Begin 12
Product Features 12
Connecting the Equipment 15
Configuration and Management of the ATA 16
Registering a Cisco SPA302D Handset 17
Additional Information 18
Using the IVR for Administration 18
Mounting the ATA 22
Elements of the User Interface 24
Chapter 2: Quick Setup for Voice over IP Service 26
Chapter 3: Configuring the Network 29
Basic Setup 29
Network Service 29
Internet Settings 31
Network Settings for the LAN and DHCP Server 34
Time Settings 37
Advanced Settings 39
Port Setting 39
MAC Address Clone 40
VPN Passthrough 41
VLAN 42
CDP & LLDP 43
Application 44
Quality of Service (QoS) 44
Port Forwarding 45

Cisco SPA232D Administrator Guide 2
Contents
Manually Adding Port Forwarding 46
DMZ 48
Chapter 4: Configuring the Voice Settings 49
Information 50
System 60
SIP 64
Provisioning 74
Regional 80
Line 1 Settings (PHONE Port) 100
Configuration Tables 117
PSTN (LINE Port) 125
User 1 147
PSTN User 152
DECT Line 1 - DECT Line 10 154
DECT User 174
SSL/TLS Certificate 177
Server Certification 177
Client Certification 178
Chapter 5: Administration Settings 179
Management 179
Web Access Management 180
TR-069 182
SNMP 184
User List (Password Management) 186
Bonjour 187
Reset Button 187
Logging 187
Log Module 188
Log Setting 190

Cisco SPA232D Administrator Guide 3
Contents
Log Viewer 192
Diagnostics 193
Ping Test 193
Traceroute Test 193
Factory Defaults 194
Firmware Upgrade 194
Configuration Management 195
Backup Configuration 195
Restore Configuration 195
Reboot 196
Chapter 6: Viewing the Status and Statistics 197
System Information 197
Interface Information 198
Internet Status 199
Port Statistics 200
Memory Information 201
DHCP Server Information 202
Optimizing Fax Completion Rates 212
VoIP-to-PSTN and PSTN-to-VoIP Calling 215
How VoIP-To-PSTN Calls Work 215
How PSTN-To-VoIP Calls Work 217
Terminating Gateway Calls 218
VoIP Outbound Call Routing 218
Configuring VoIP Failover to PSTN 220
Sharing One VoIP Account Between the PHONE and LINE Ports 220
PSTN Call to Ring Line 1 221
Symmetric RTP 221
Call Progress Tones 221
Call Scenarios 222
PSTN to VoIP Call with and Without Ring-Thru 222

Cisco SPA232D Administrator Guide 4
Contents
VoIP to PSTN Call With and Without Authentication 223
Call Forwarding to PSTN Gateway 224
Configuring Dial Plans 225
Digit Sequences 226
Acceptance and Transmission of the Dialed Digits 230
Dial Plan Timer (Off-Hook Timer) 231
Interdigit Long Timer (Incomplete Entry Timer) 232
Interdigit Short Timer (Complete Entry Timer) 233
Resetting the Control Timers 234

1
Cisco SPA232D Administration Guide 5
Getting Started
Thank you for choosing the Cisco SPA232D Mobility Enhanced Analog Telephone
Adapter (ATA). This chapter provides more information about the features of the
product and provides instructions about connecting the equipment and getting
started in the web-based configuration utility.
Feature Overview
With the Cisco SPA232D Mobility Enhanced ATA, you can provide your analog
phone and Cisco SPA302D cordless handsets with access to analog and Internet
phone services through a standard RJ-11 phone port and a built-in DECT base
station. The ATA supports up to five Cisco SPA302D handsets. The ATA connects
to the Internet through a broadband (DSL or cable) modem or router. The ATA can
be used with an on-site call-control system or an Internet-based call-control
system.
The ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables
carrier-class residential and business IP Telephony services delivered over
broadband or high-speed Internet connections. An ATA maintains the state of each
call it terminates and reacts appropriately to user input events (such as on/off hook
or hook flash). The ATAs use the Session Initiation Protocol (SIP) open standard so
there is little or no involvement by a “middle-man” server or media gateway
controller. SIP allows inter-operation with all ITSPs that support SIP.
The system supports four simultaneous calls, including “active” calls and “on-hold”
calls. A phone or DECT handset can handle one on-hold call and one active call
simultaneously.

Getting Started
Feature Overview
Cisco SPA232D Administration Guide 6
1
Understanding Voice Service Operations
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 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.
ATA Voice Features
The ATA can be custom provisioned within a wide range of configuration
parameters. The following sections describe the factors that contribute to voice
quality:
•Supported Codecs
•SIP Proxy Redundancy
•Other ATA Voice Features
IP
Analogue
Phone
(FXS port)
or
SPA232D typical topology
or
FXS/DECT
LAN
PC
FXO
WAN
Active PSTN L
Cisco SPA232D
Internet Service Provider
Internet (ADSL/Cable
Modem)
Cordless
Phone
(DECT)
Fax Machine
(FXS port)

Getting Started
Feature Overview
Cisco SPA232D Administration Guide 7
1
Supported Codecs
The ATA supports the codecs listed below. You can use the default settings or
configure the codec settings in the Audio Configuration section of these pages:
Line 1 Settings (PHONE Port), PSTN (LINE Port), and DECT Line 1 - DECT Line
10.
Codec Description
G.711 (A-law and mu-law) Very low complexity codecs that support
uncompressed 64 kbps digitized voice
transmissions at one through ten 5 ms voice frames
per packet. These codecs provide the highest
narrow-band voice quality and uses the most
bandwidth of any of the available codecs.
G.726-32 Low complexity codec that supports compressed
32 kbps digitized voice transmission at one through
ten 10 ms voice frames per packet. This codec
provides high voice quality.
G.729a ITU G.729 voice coding algorithm used to
compress digitized speech. G.729a is a reduced
complexity version of G.729 requiring about half
the processing power of G.729. The G.729 and
G.729a bit streams are compatible and
interoperable, but not identical.

Getting Started
Feature Overview
Cisco SPA232D Administration Guide 8
1
SIP Proxy Redundancy
In typical commercial IP Telephony deployments, all calls are established through
a SIP proxy server. A typical SIP proxy server can handle thousands of
subscribers. It is important that a backup server be available so that an active
server can be temporarily switched out for maintenance. The ATA supports the
use of backup SIP proxy servers (through DNS SRV) so that service disruption is
minimized.
An easy way to support proxy redundancy is to configure your DNS server with a
list of SIP proxy addresses. The ATA can be instructed to contact a SIP proxy
server in a domain named in the SIP message. The ATA consults the DNS server to
get a list of hosts in the given domain that provide SIP services. If an entry exists,
the DNS server returns an SRV record that contains a list of SIP proxy servers for
the domain, with their host names, priority, listening ports, and so on. The ATA tries
to contact the list of hosts in the order of their stated priority.
If the ATA is currently using a lower priority proxy server, it periodically probes the
higher priority proxy to see whether it is online, and switches back to the higher
priority proxy when possible. You can use the default settings or configure the
Proxy Redundancy Method in the Proxy and Registration section of the Line 1
Settings (PHONE Port) page and the DECT Line 1 - DECT Line 10 pages.
Other ATA Voice Features
•Silence Suppression and Comfort Noise Generation
Voice Activity Detection (VAD) with Silence Suppression is a means of
increasing the number of calls supported by the network by reducing the
average bandwidth required for a single call. VAD distinguishes between
speech and non-speech signals, and Silence Suppression removes the
natural silences that occur in a conversation. Therefore the IP bandwidth is
used only to transmit speech. Comfort Noise Generation provides
artificially-generated background white noise (sounds) to reassure callers
that their calls are still connected during the silent periods. You can enable
this feature in the Audio Configuration section of these pages: Line 1
Settings (PHONE Port), PSTN (LINE Port), and DECT Line 1 - DECT Line
10.
•Modem and Fax Pass-Through
-Modem pass-through mode can be triggered by predialing the Vertical
Service Activation Code for the Modem Line Toggle Code. You can
configure this setting in the Vertical Service Activation Codes section
of the Regional page.

Getting Started
Feature Overview
Cisco SPA232D Administration Guide 9
1
-FAX pass-through mode is triggered by the detection of a CED/CNG
tone or an NSE event.
-Echo canceller is automatically disabled for Modem passthrough mode.
-Echo canceller is disabled for FAX pass-through if the parameter FAX
Disable ECAN (Line 1 or 2 tab) is set to “yes” for that line (in that case
FAX pass-through is the same as Modem pass-through)
-Call waiting and silence suppression are automatically disabled for both
FAX and Modem pass-through. In addition, out-of-band DTMF
transmission is disabled during modem or fax passthrough.
NOTE For more information, please refer to Audio Configuration, page 113 and
Configuration Tables, page 117.
•Adaptive Jitter Buffer
The ATA can buffer incoming voice packets to minimize the impact of
variable network delays. This process is known as jitter buffering. The size
of the jitter buffer adjusts to changing network conditions. The ATA has a
Network Jitter Level control setting for each line of service. The jitter level
determines how aggressively the ATA tries to shrink the jitter buffer over
time to achieve a lower overall delay. If the jitter level is higher, it shrinks
more gradually. If jitter level is lower, it shrinks more quickly. You can use the
default settings or configure this feature in the Network Settings section of
these pages: Line 1 Settings (PHONE Port), PSTN (LINE Port), and DECT
Line 1 - DECT Line 10.
•Adjustable Audio Frames Per Packet
This feature allows the user to set the number of audio frames contained in
one RTP packet. Packets can be adjusted to contain from 1–10 audio
frames. Increasing the number of packets decreases the bandwidth utilized,
but it also increases delay and may affect voice quality. You can configure
this setting in the RTP Parameters section of the SIP page.
•DTMF Relay
The ATA may relay DTMF digits as out-of-band events to preserve the
fidelity of the digits. This can enhance the reliability of DTMF transmission
required by many IVR applications such as dial-up banking and airline
information. You can configure this setting in the RTP 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 that an end user is advised of status (such
as ringback) Parameters for each type of tone (for instance a dial tone