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setting value is the same as Account Name, it is not necessary

for you to check the box and set any value in this field.

Password

The password provided to you when you registered with a SIP

service.

Expiry Time

The time duration that your SIP Registrar server keeps your

registration record. Before the time expires, the router will send

another register request to SIP Registrar again.

NAT Traversal Support

If the router (e.g.

,

broadband router) you use connects to

internet by other device, you have to set this function for your

necessity.

None –

Disable this function.

Stun

– Choose this option if there is Stun server provided for

your router.

Manual

– Choose this option if you want to specify an external

IP address as the NAT transversal support.

Nortel

– If the soft-switch that you use supports Nortel solution,

you can choose this option.

Ring Port

Set Phone1 or Phone2 as the default ring port for this SIP

account.

Ring Pattern

Choose a ring tone type for the VoIP phone call.

3.5.3 Phone Settings

This page allows user to set phone settings for Phone 1 and Phone 2 respectively. However, it

changes slightly according to different model you have.

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Phone List

Port – t

here are two phone ports provided here for you to

configure.

Phone1/Phone2

allow you to set general settings for

PSTN phones.

Call Feature

– A brief description for call feature will be

shown in this field for your reference.

Codec

– The default Codec setting for each port will be shown

in this field for your reference. You can click the number below

the Index field to change it for each phone port.

Tone

- Display the tone settings that configured in the advanced

settings page of Phone Index.

Gain

- Display the volume gain settings for Mic/Speaker that

configured in the advanced settings page of Phone Index.

Default SIP Account

– “draytel_1” is the default SIP account.

You can click the number below the Index field to change SIP

account for each phone port.

DTMF Relay

– Display DTMF mode that configured in the

advanced settings page of Phone Index.

RTP

Symmetric RTP

– Check this box to invoke the function. To

make the data transmission going through on both ends of local

router and remote router not misleading due to IP lost (for

example, sending data from the public IP of remote router to the

private IP of local router), you can check this box to solve this

problem.

Dynamic RTP Port Start

- Specifies the start port for RTP

stream. The default value is 10050.

Dynamic RTP Port End

- Specifies the end port for RTP

stream. The default value is 15000.

RTP TOS

– It decides the level of VoIP package. Use the drop

down list to choose any one of them.

Detailed Settings for Phone Port

Click the number link for Phone port, you can access into the following page for configuring

Phone settings.

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Hotline

Check the box to enable it. Type in the SIP URL in the field for

dialing automatically when you pick up the phone set.

Session Timer

Check the box to enable the function. In the limited time that

you set in this field, if there is no response, the connecting call

will be closed automatically.

Call Forwarding

There are four options for you to choose.

Disable

is to close call

forwarding function.

Always

means all the incoming calls will

be forwarded into SIP URL without any reason.

Busy

means

the incoming calls will be forwarded into SIP URL only when

the local system is busy.

No Answer

means if the incoming

calls do not receive any response, they will be forwarded to the

SIP URL by the time out.

SIP URL

– Type in the SIP URL (e.g., [email protected] or

[email protected]) as the site for call forwarded.

Time Out

– Set the time out for the call forwarding. The

default setting is 30 sec.

DND (Do Not Disturb)

mode

Set a period of peace time without disturbing by VoIP phone

call. During the period, the one who dial in will listen busy

tone, yet the local user will not listen any ring tone.

Index (1-15) in Schedule -

Enter the index of schedule profiles

to control the DND mode according to the preconfigured

schedules. Refer to section

3.8.2 Schedule

for detailed

configuration.

Index (1-60) in Phone Book -

Enter the index of phone book

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63

profiles. Refer to section

VoIP>>DialPlan>>Phone Book

for

detailed configuration.

CLIR (hide caller ID)

Check this box to hide the caller ID on the display panel of the

phone set.

Call Waiting

Check this box to invoke this function. A notice sound will

appear to tell the user new phone call is waiting for your

response. Click hook flash to pick up the waiting phone call.

Call Transfer

Check this box to invoke this function. Click hook flash to

initiate another phone call. When the phone call connection

succeeds, hang up the phone. The other two sides can

communicate, then.

Prefer Codec

Select one of five codecs as the default for your VoIP calls. The

codec used for each call will be negotiated with the peer party

before each session, and so may not be your default choice. The

default codec is G.729A/B; it occupies little bandwidth while

maintaining good voice quality.

If your upstream speed is only 64Kbps, do not use G.711 codec.

It is better for you to have at least 256Kbps upstream if you

would like to use G.711.

Single Codec

– If the box is checked, only the selected Codec

will be applied.

Packet Size

-The amount of data contained in a single packet.

The default value is 20 ms, which means the data packet will

contain 20 ms voice information.

Voice Active Detector -

This function can detect if the voice

on both sides is active or not. If not, the router will do

something to save the bandwidth for other using. Click On to

invoke this function; click off to close the function.

Default SIP Account

You can set SIP accounts (up to six groups) on SIP Account

page. Use the drop down list to choose one of the profile names

for the accounts as the default one for this phone setting.

Play dial tone only when account registered -

Check this box

to invoke the function.