DrayTek VigorPro 5300 Router Manual PDF (Setup & Configuration Guide)

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Any

as ring port for specified SIP account. By the way,

ISDN-S0 can be used by mapping with MSN numbers.

Ring Pattern

Choose a ring tone type for the VoIP phone call.

3.12.3 Phone Settings

This page allows user to set phone settings for Phone1, Phone2, ISDN1-S0 and

ISDN2-TE/S0 respectively.

Phone List

Port –

There are four phone ports provided here for you to

configure. Three (Index 1 to 3) are fixed and one (Index 4) is

configurable.

Phone1

and

Phone2

allow you to set general

settings for PSTN phones.

ISDN1-S0

and

ISDN2-TE

allow

you to set common settings for ISDN network connection.

ISDN2 port is configurable. Please use the drop down list to

choose

ISDN2-TE

for Internet connection or choose

ISDN2-S0

(ISDN intern) for ISDN phone. In addition, you

can connect six phones to this router in certain case. Please

refer to

Section 4-1

for detailed information of ISDN

phone/network connection.

If you want to enable function of ISDN On-Net/Off-Net, you

have to choose ISDN2-TE.

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.

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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 Phone1/Phone2 Port

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

Phone settings. Below is the sample page for Phone1.

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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

3.5.2 Schedule

for

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detailed configuration.

Index (1-60) in Phone Book -

Enter the index of phone book

profiles. Refer to section

3.11.1 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.

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Play dial tone only when account registered -

Check this

box to invoke the function.

Default Call Route

It determines the default direction for the call route of the

router.

To ISDN (for VoIP)

- The router is set by using ISDN call.

To change ISDN call into VoIP call, please dial the character

in this field for transferring. The character that you can type

can be *, #, and 0~9.

To VoIP (for ISDN)

- The router is set by using VoIP call. To

change VoIP call into ISDN call, please dial the character in

this field for transferring. The character that you can type can

be *, #, and 0~9.

In addition, you can press the

Advanced

button to configure tone settings, volume gain,

MISC and DTMF mode.

Advanced

setting is provided for fitting the telecommunication

custom for the local area of the router installed. Wrong tone settings might cause

inconvenience for users. To set the sound pattern of the phone set, simply choose a proper

region to let the system find out the preset tone settings and caller ID type automatically. Or

you can adjust tone settings manually if you choose User Defined. TOn1, TOff1, TOn2 and

TOff2 mean the cadence of the tone pattern. TOn1 and TOn2 represent sound-on; TOff1 and

TOff2 represent the sound-off.

Region

Select the proper region which you are located. The common

settings of

Caller ID Type

,

Dial tone

,

Ringing tone

,

Busy

tone

and

Congestion tone

will be shown automatically on the

page. If you cannot find out a suitable one, please choose

User Defined

and fill out the corresponding values for dial

tone, ringing tone, busy tone, congestion tone by yourself for

VoIP phone.

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