ZyXEL P-2002 User Guide - Page 63

P-2002 Series User’s Guide

Chapter 6 VoIP Screens

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Server Port

Enter the STUN server’s listening port for STUN requests in this field. Leave this

field set to the default if your VoIP service provider did not give you a server port

number for STUN.

Use NAT

Enable this feature to use a NAT router’s public IP address and SIP port number in

the Prestige’s SIP messages.

You must also configure the NAT router to forward traffic with this port number to

the Prestige.

This eliminates the need for STUN or a SIP ALG.

IP Address

Enter the NAT router’s public IP address or domain name (up to 127 ASCII

characters) in this field.

Server Port

Enter the port number that your SIP sessions use with the public IP address of the

NAT router.

Outbound Proxy

Enable this feature

if your VoIP service provider has a SIP outbound server to

handle voice calls. This allows the Prestige to work with any type of NAT router and

eliminates the need for STUN or a SIP ALG. Turn off a SIP ALG on a NAT router in

front of the Prestige to keep it from retranslating the IP address (since this is

already handled by the outbound proxy server).

Server Address

Enter the IP address or domain name (up to 127 ASCII characters) of the SIP

outbound proxy server in this field.

Server Port

Enter the SIP outbound proxy server’s listening port for SIP outbound proxy

requests in this field. Leave this field set to the default if your VoIP service provider

did not give you a server port number for the SIP outbound proxy server.

Enable NAT Keep

Alive

You must have outbound proxy enabled to use NAT keep alive.

Enable NAT keep alive to have the Prestige send SIP notify messages to the SIP

server. Use this to keep a NAT router located between the Prestige and the SIP

server from timing out and dropping your Prestige’s SIP NAT sessions.

Keep Alive Interval

Set how often (in seconds) the Prestige should send SIP notify messages to the

SIP server.

DTMF Mode

The Dual-Tone MultiFrequency (DTMF) mode sets how the Prestige handles the

tones that your telephone makes when you push its buttons. It is recommended

that you use the same mode that your VoIP service provider uses.

Select

RFC 2833

to send the DTMF tones in RTP packets.

Select

PCM

(Pulse Code Modulation) to include the DTMF tones in the voice data

stream. This method works best when you are using a codec that does not use

compression (like G.711). Codecs that use compression (like G.729) could distort

the tones.

Select

SIP INFO

to send the DTMF tones in SIP messages.

MWI (Message

Waiting Indication)

Enable Message Waiting Indication (MWI) to have your phone give you a

message–waiting (beeping) dial tone when you have a voice message(s). Your

voice service provider must have a messaging system that supports this feature.

Expiration Time

Use this field to set how long the SIP server should continue providing the

message waiting service after receiving a SIP SUBSCRIBE message from the

Prestige.

The SIP server stops providing the message waiting service if it has not received

another SIP SUBSCRIBE message from the Prestige before this time period

expires.

Table 14

VoIP Advanced (continued)

LABEL

DESCRIPTION

Section	Page
P-2002 Series	1
Copyright	3
Federal Communications Commission (FCC) Interference Statement	4
Safety Warnings	6
ZyXEL Limited Warranty	7
Customer Support	8
Table of Contents	11
List of Figures	17
List of Tables	19
Preface	21
Introducing the Prestige	23
1.1 Prestige VoIP Analog Telephone Adaptor Overview	23
1.2 Prestige 2002L VoIP Analog Telephone Adaptor with Lifeline	23
1.3 Features	23
1.4 LEDs	25
1.5 Applications	26
1.5.1 Make Calls via Internet Telephony Service Provider	26
1.5.2 Make Calls via IP-PBX	27
1.5.3 Make Peer-to-peer Calls	27
Introducing the Web Configurator	29
2.1 Web Configurator Overview	29
2.2 Accessing the Prestige Web Configurator	29
2.3 Resetting the Prestige	31
2.3.1 Procedure To Use The Reset Button	31
2.4 Navigating the Prestige Web Configurator	31
2.5 Common Screen Command Buttons	33
System Screens	35
3.1 System Overview	35
3.2 General Setup	35
3.2.1 Domain Name	35
3.3 General Setup Configuration	35
3.4 Configuring Password	36
3.5 Pre-defined NTP Time Servers List	37
3.6 Configuring Time Setting	37
Ethernet Screens	41
4.1 ETHERNET Overview	41
4.1.1 IP Address and Subnet Mask	41
4.1.2 DNS Server Address Assignment	41
4.1.3 PPPoE Encapsulation	42
4.2 Configuring Ethernet	42
4.3 Configuring the Ethernet Management Port	44
Introduction to VoIP	45
5.1 Introduction to VoIP	45
5.2 Introduction to SIP	45
5.2.1 SIP Identities	45
5.2.1.1 SIP Number	45
5.2.1.2 SIP Service Domain	45
5.2.2 SIP Call Progression	46
5.2.3 SIP Servers	46
5.2.3.1 SIP User Agent	46
5.2.3.2 SIP Proxy Server	47
5.2.3.3 SIP Redirect Server	47
5.2.3.4 SIP Register Server	48
5.2.4 RTP	48
5.3 NAT	48
5.3.1 NAT Example	49
5.3.2 NAT Types	49
5.3.2.1 Full Cone NAT	50
5.3.2.2 Restricted Cone NAT	51
5.3.2.3 Port Restricted Cone NAT	52
5.3.2.4 Symmetric NAT	52
5.4 NAT and SIP	53
5.4.1 SIP ALG	53
5.4.2 Use NAT Address on SIP and RTP	53
5.4.3 STUN	54
5.4.4 Outbound Proxy	54
5.5 Pulse Code Modulation	54
5.6 Voice Coding	54
5.6.1 G.711	55
5.6.2 G.729	55
5.7 PSTN Call Setup Signaling	55
5.8 MWI (Message Waiting Indication)	55
VoIP Screens	57
6.1 VoIP Introduction	57
6.2 VoIP Configuration	57
6.3 Custom Tones (IVR)	59
6.3.0.1 Recording Custom Tones	60
6.3.0.2 Listening to Custom Tones	60
6.3.0.3 Deleting Custom Tones	60
6.4 Advanced VoIP Settings Configuration	60
6.5 Quality of Service (QoS)	64
6.5.1 Type Of Service (ToS)	64
6.5.2 DiffServ	64
6.5.2.1 DSCP and Per-Hop Behavior	65
6.5.3 VLAN	65
6.6 QoS Configuration	65
Phone	67
7.1 Phone Introduction	67
7.1.1 Voice Activity Detection/Silence Suppression	67
7.1.2 Comfort Noise Generation	67
7.1.3 Echo Cancellation	67
7.2 Phone Port Configuration	67
7.3 Supplementary Phone Services Overview	69
7.3.1 The Flash Key	69
7.3.2 Europe Type Supplementary Phone Services	69
7.3.2.1 European Call Hold	70
7.3.2.2 European Call Waiting	70
7.3.2.3 European Call Transfer	71
7.3.2.4 European Three-Way Conference	71
7.3.3 USA Type Supplementary Services	71
7.3.3.1 USA Call Hold	71
7.3.3.2 USA Call Waiting	72
7.3.3.3 USA Call Transfer	72
7.3.3.4 USA Three-Way Conference	72
7.4 Common Phone Configuration	72
Phone Book	75
8.1 Phone Book Introduction	75
8.1.1 Speed Dial	75
8.1.1.1 Peer-to-Peer Calls	75
8.1.2 Lifeline (Prestige 2002L)	75
8.2 Speed Dial Configuration	75
8.3 Call Forward	77
8.4 Lifeline Configuration (Prestige 2002L)	80
Logs	81
9.1 Viewing Logs	81
9.2 Log Message Descriptions	82
Maintenance	85
10.1 Maintenance Overview	85
10.2 Status Screen	85
10.3 F/W Upload Screen	86
10.4 Configuration Screen	88
10.4.1 Backup Configuration	89
10.4.2 Restore Configuration	89
10.4.3 Back to Factory Defaults	90
10.5 Restart Screen	91
Phone Usage	93
11.1 Dialing a Telephone Number	93
11.2 Using Speed Dial to Dial a Telephone Number	93
11.3 Internal Calls	93
11.4 Checking the Prestige’s IP Address	93
11.5 Auto Firmware Upgrade	94
Troubleshooting	95
12.1 Problems Starting Up the Prestige	95
12.2 Problems with the LAN or PC LED	95
12.3 Problems with the LAN Interface	96
12.4 Problems with Internet Access	96
12.5 Problems with the Web Configurator	97
12.6 Problems with the Password	97
12.7 Problems with Telephone or Telephone Port	98
12.8 Problems with Voice Service	98
12.9 Pop-up Windows, JavaScripts and Java Permissions	99
12.9.1 Internet Explorer Pop-up Blockers	99
12.9.1.1 Disable Pop-up Blockers	99
12.9.1.2 Enable Pop-up Blockers with Exceptions	100
12.9.2 JavaScripts	102
12.9.3 Java Permissions	104
12.9.3.1 JAVA (Sun)	105
Product Specifications	107
Setting up Your Computer’s IP Address	111
IP Subnetting	123
PPPoE	131
Internal SPTGEN	133
Index	161
Numerics	161
A	161
B	161
C	161
D	162
E	162
F	162
G	163
H	163
I	163
J	163
K	163
L	163
M	164
N	164
O	164
P	164
Q	165
R	165
S	166
T	167
U	167
V	167
W	167

ZyXEL P-2002 User Guide - Page 63

P-2002 Series User's Guide, VoIP Screens, VoIP Advanced continued

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