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Page 56 / 95 Scroll up to view Page 51 - 55
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
Page 57 / 95
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
Page 58 / 95
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
Page 59 / 95
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
Page 60 / 95
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.

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