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

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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.

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.

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Detailed Settings for ISDN1-S0 Port

Click the number link of Index 3 (ISDN1-S0), you can access into the following page for

configuring Phone settings.

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.

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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

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.

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

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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.

ISDN-S0 port can pick up multiple incoming calls

simultaneously. Therefore different phone sets (MSN30 to

MSN39) can use different SIP accounts to call out through

this port.

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, DTMF mode and MSN number.

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.

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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.

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

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