Cisco SPA232D Manuale di istruzioni

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

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

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

Analogue

Phone

(FXS port)

SPA232D typical topology

FXS/DECT

LAN

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

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

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

-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

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