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

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VigorPro5300 Series User’s Guide

173

default setting.

Ring Frequency

- This setting is used to drive the frequency

of the ring tone. It is recommended for you to use the default

setting.

DTMF

DTMF Mode –

There are four DTMF modes for you to

choose.

InBand

-

Choose this one then the Vigor will send the DTMF

tone as audio directly when you press the keypad on the phone

OutBand

-

Choose this one then the Vigor will capture the

keypad number you pressed and transform it to digital form

then send to the other side; the receiver will generate the tone

according to the digital form it receive. This function is very

useful when the network traffic congestion occurs and it still

can remain the accuracy of DTMF tone.

SIP INFO

-

Choose this one then the Vigor will capture the

DTMF tone and transfer it into SIP form. Then it will be sent

to the remote end with SIP message.

Payload Type (rfc2833)

- Choose a number from 96 to 127,

the default value was 101. This setting is available for the

OutBand (RFC2833) mode.

Detailed Settings for ISDN2-TE Port (Available for

VSn

model only)

Vigor2930VSn allows users to switch the function of ISDN2 port between TE or S0 mode.

Please use the drop down list to choose the one you want.

If you choose ISDN2-S0, please refer to Detailed Settings for Phone1, Phone2, ISDN1-S0

for the configuration. However, if you choose ISDN-TE and click the number link for that

port, you will see the following page.

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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

detailed configuration.

Index (1-60) in Phone Book -

Enter the index of phone book

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profiles. Refer to section

3.10.1 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.

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.

FXO Feature

Enable ISDN to VoIP (On-Net) Calls

– Check this box to

make all the outgoing calls from ISDN line to be forwarded to

receivers by Internet.

Enable VoIP to ISDN (Off-Net) Calls

–Check this box to

make all the incoming calls coming from Internet to be

forwarded to receivers by ISDN line.

Loop Through to Phone Port

– Choose this radio button to

make all the calls controlled by traditional PSTN phone. It

will tack effect only if MSN mapping ring port is not

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configured. In addition, you can specify which port (both

phone 1 and phone 2, phone 1 only or phone 2 only) will ring.

Loop Through to ISDN1-S0 Port

– Choose this radio button

to make all the calls controlled by ISDN line.

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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Also, you can specify each field for your necessity. It is

recommended for you to use the default settings for VoIP

communication.

Volume Gain

Mic Gain (1-10)/Speaker Gain (1-10)

- Adjust the volume of

microphone and speaker by entering number from 1- 10. The

larger of the number, the louder the volume is.

MISC

Dial Tone Power Leve

l - This setting is used to adjust the

loudness of the dial tone. The smaller the number is, the

louder the dial tone is.

It is recommended for you to use the

default setting.

Authentication PIN Code

Check for ISDN to VoIP Calls

– Set a pin code for the router

to authenticate which one is allowed to dial ISDN to VoIP call.

The figure that you can type in this field is limited from three

to eight with digits from zero to nine.

Check for VoIP to ISDN Calls

- Set a pin code for the router

to authenticate which one is allowed to dial VoIP to ISDN call.

The figure that you can type in this field is limited from three

to eight with digits from zero to nine.

DTMP

DTMF mode –

There are four selections provided here:

InBand:

Choose this one then the Vigor will send the DTMF

tone as audio directly when you press the keypad on the phone

OutBand:

Choose this one then the Vigor will capture the

keypad number you pressed and transform it to digital form

then send to the other side; the receiver will generate the tone

according to the digital form it receive. This function is very

useful when the network traffic congestion occurs and it still

can remain the accuracy of DTMF tone.

SIP INFO:

Choose this one then the Vigor will capture the

DTMF tone and transfer it into SIP form. Then it will be sent

to the remote end with SIP message.

Payload Type (rfc2833) -

Choose a number from 96 to 127,

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