EG101

User’s Guide

50



Enter

UID (User Name)

for Line 1



Enter the

Password



Enter the

Display Name

which will be shown in the called party when you call out.



Enter the

Login ID

if your login username is not the UID.



Enter the

Primary Proxy Server

IP address or domain name and also port

number



Enter the

Second Proxy Server

IP address or domain name and also port

number is applicable



Enter the

Registrar Server

IP address or domain name and also port number if

applicable.



Enter the

Outbound Proxy Server

IP address or domain name and also port

number if applicable.



Enter the

Register Expiry Number

of seconds which the device will try to register

in SIP server during the time period.



Select the

NAT Keep-alive Method

from the list



Enter the

SIP Proxy Require

if necessary



Select the

User Prefered Audio Codec

from the list



Enter the

RTP Base number



Check to enable

Use STUN for NAT Mapping to pass through NAT



Check to enable

VIA rport



Check to enable

DNS SVR



Click

Save

to save the configuration

Line 2 Setting

This page allows you to setup the parameters of

Line 1

including username, password,

codec, and call features.

Figure 58: VoIP Configuration – Line 2 Setting

EG101

User’s Guide

51

Global Setting:



Select to Enable this line or not



Enter

UID (User Name)

for Line 2



Enter the

Password



Enter the

Display Name

which will be shown in the called party when you call out.



Enter the

Login ID

if your login username is not the UID.



Enter the

Primary Proxy Server

IP address or domain name and also port

number



Enter the

Second Proxy Server

IP address or domain name and also port

number is applicable



Enter the

Registrar Server

IP address or domain name and also port number if

applicable.



Enter the

Outbound Proxy Server

IP address or domain name and also port

number if applicable.



Enter the

Register Expiry Number

of seconds which the device will try to register

in SIP server during the time period.



Select the

NAT Keep-alive Method

from the list



Enter the

SIP Proxy Require

if necessary



Select the

User Prefered Audio Codec

from the list



Enter the

RTP Base number



Check to enable

Use STUN for NAT Mapping to pass through NAT



Check to enable

VIA rport



Check to enable

DNS SVR



Click

Save

to save the configuration

EG101

User’s Guide

52

RTP/Codec Setup

This page allows you to setup the parameters of Real Time Protocol (RTP) and voice

codec to control the quality of voice connection.

Figure 59: VoIP Configuration – RTP/Codec Setup

Global Setting:



Enter the

TOS

byte. The TOS stands for Type of Service. It is defined in the

RFC1394 and used in RTP packet.



Enter the

RTP Packet Period

of milliseconds. Suggest leave it as default for

better quality.



Select the

DTMF Method

from the list. The device provides in-band DFTM

signaling and out-band DTMF signaling.



Enter the

DTMP PayLoad Type

, the default is 101.



Select the

Jitter Buffer

and Echo Cancellation from the list. Suggest leave it as

default for better quality.



Select the

Fax Codec

from the list and check to

Enable Fax Pass Through

and

Enable CED Tone Detection

.



Check to select the available Audio Codec from the list.



Click

Save

to save the configuration

EG101

User’s Guide

53

Operational Setup

This page allows you to configure the call features including call forward, call waiting,

three-way conference and so on as well as tones, FXS and caller ID.

Figure 60: VoIP Configuration – Operational Setup 1

EG101

User’s Guide

54

Figure 61: VoIP Configuration – Operational Setup 2

Global Setting:



Place a check in the list of call features which is supported in Line 1 and Line 2.



Enter the value of feature timers including Redialing Duration, Retrial Interval,

OnHook Delay, PSTN Session Progress Timeout, Call Waiting Ring Timeout.



Enter the value of Signal Timers including Call Waiting Period, Reorder Delay,

Ring Timeout, No Answer Timeout, Min. Hook Flash Time, Max. Hook Flash

Time.



Place a check in the list of Operational Flags including Enable Call Forwarding

on Server, Enable Call Return on Server, Allow Incoming Call to Phone 1, Allow

Incoming Call to Phone 2, CLIR Method (Anonymous Form or Use Privacy

Header), Phone 1 Use (SIP1 or SIP2), Phone 2 Use (SIP1 or SIP2)



Define the Service Activation Code for each call service



Configure your own Dial Plan.



Click

Save

to save the configuration

Those parameters are related to the VoIP service provided by Voice Service Provider.

Please leave it as default or consult technician in advance to understand and

configure it.