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Problems with Telephone or Telephone Port, Problems with Voice Service

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P-2002 Series User's Guide 12.7 Problems with Telephone or Telephone Port Table 40 Troubleshooting Telephone PROBLEM CORRECTIVE ACTION There is no dial tone or I can't make or receive calls. or There is beeping instead of the dial tone. Check the telephone connections and telephone wire. Beeping means that there is not a SIP account registered for the phone to use. You can check the Prestige's IP addresses and VoIP status in the Maintenance Status screen. Make sure you have the VoIP screen properly configured. If you configured a SIP account to receive calls on only one of the phone ports, make sure your phone is connected to that port. Make sure you have the Phone Port screen properly configured. If you configured a phone port to only use one of the SIP accounts for outgoing calls, make sure that SIP account is properly configured and active (see the VoIP and Maintenance Status screens). There is a beep A single beep before the dial tone indicates that there is a voice message for SIP before the dial tone. account 1. Two beeps before the dial tone indicate that there is a voice message for SIP account 2. Use your voice service provider's instructions to check your voice messages. 12.8 Problems with Voice Service Table 41 Troubleshooting Voice Service PROBLEM CORRECTIVE ACTION After the VoIP is configured and working, others are unable to call you or you lose your connection during a call. There is a NAT router between the Prestige and the SIP server. This could be caused by a short NAT UDP session timeout on the NAT router. When the SIP session's entry in the NAT table times out, the NAT router does not have any record to use for forwarding VoIP traffic to the Prestige. If possible, set the NAT router to use a longer NAT UDP session timeout. Otherwise, try one of the following: • Shorten the registration expiration period (see the Expiration Duration field in the VoIP Advanced screen) in order to cause the Prestige to re-register with the SIP register server more frequently. Note that this will not help if the SIP register server enforces a long registration expiration period (since the Prestige will also use the period set by the SIP register server). • Use STUN. If your VoIP service provider does not have a STUN server, you can still enable STUN and enter the IP address and port number of the SIP server in the STUN server fields. This causes the Prestige to send STUN requests to the SIP server. While this will not make STUN work (since there won't be any responses to the STUN requests), it should keep the NAT UDP session in the NAT router. 98 Chapter 12 Troubleshooting

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

Match case Limit results 1 per page

P-2002 Series User’s Guide

Chapter 12 Troubleshooting

12.7

Problems with Telephone or Telephone Port

12.8

Problems with Voice Service

Table 40

Troubleshooting Telephone

PROBLEM

CORRECTIVE ACTION

There is no dial

tone or I can’t make

or receive calls.

There is beeping

instead of the dial

tone.

Check the telephone connections and telephone wire.

Beeping means that there is not a SIP account registered for the phone to use.

You can check the Prestige’s IP addresses and VoIP status in the

Maintenance

Status

screen.

Make sure you have the

VoIP

screen properly configured. If you configured a SIP

account to receive calls on only one of the phone ports, make sure your phone is

connected to that port.

Make sure you have the

Phone Port

screen properly configured. If you

configured a phone port to only use one of the SIP accounts for outgoing calls,

make sure that SIP account is properly configured and active (see the

VoIP

and

Maintenance Status

screens).

There is a beep

before the dial tone.

A single beep before the dial tone indicates that there is a voice message for SIP

account 1.

Two beeps before the dial tone indicate that there is a voice message for SIP

account 2.

Use your voice service provider’s instructions to check your voice messages.

Table 41

Troubleshooting Voice Service

PROBLEM

CORRECTIVE ACTION

After the VoIP is

configured and

working, others are

unable to call you

or you lose your

connection during a

call. There is a NAT

router between the

Prestige and the

SIP server.

This could be caused by a short NAT UDP session timeout on the NAT router.

When the SIP session’s entry in the NAT table times out, the NAT router does not

have any record to use for forwarding VoIP traffic to the Prestige.

If possible, set the NAT router to use a longer NAT UDP session timeout.

Otherwise, try one of the following:

•

Shorten the registration expiration period (see the

Expiration Duration

field

in the

VoIP Advanced

screen) in order to cause the Prestige to re-register

with the SIP register server more frequently. Note that this will not help if the

SIP register server enforces a long registration expiration period (since the

Prestige will also use the period set by the SIP register server).

•

Use STUN. If your VoIP service provider does not have a STUN server, you

can still enable STUN and enter the IP address and port number of the SIP

server in the STUN server fields. This causes the Prestige to send STUN

requests to the SIP server. While this will not make STUN work (since there

won’t be any responses to the STUN requests), it should keep the NAT UDP

session in the NAT router.