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SIP Advanced

The SIP Advanced page allows you to configure settings that your VoIP service provider has enabled on your SIP account and if

you have the appropriate call features and other functionality on your cordless or corded phone handsets.

Figure 41: VoIP - Advanced - Service Provider

OPTION

DEFINITION

Line

Displays the line number you want to configure

Call Waiting

Select this option for your phone if your VoIP Service Provider has enabled Call Waiting on your SIP account.

Unconditionally Call forwarding

number

Select this option if your VoIP Service Provider has enabled Call Forwarding on your SIP account and you wish

to use this feature.

Busy Call Forwarding Number

Enter the phone number to forward a call to if it arrives while the line is busy.

No Answer Call forwarding

number

Enter the phone number to forward a call to if the call is not answered.

Forward On “busy”

Select this option if your VoIP Service Provider has enabled Call Forwarding on your SIP account and you wish

to use this feature.

Forward On “No Answer”

Select this option if your VoIP Service Provider has enabled Call Forwarding on your SIP account and you wish

to use this feature.

MWI (Message Waiting

Indicator)

Select this option if your VoIP Service Provider has enabled MWI (Message Waiting Indicator) on your SIP

account and you wish to use this feature.

Anonymous Call Blocking

Select this option if your VoIP Service Provider has enabled Anonymous Call Blocking on your SIP account and

you wish to use this feature.

Anonymous Calling

Select this option if your VoIP Service Provider has enabled Anonymous Calling on your SIP account and you

wish to use this feature.

Anonymous calling mode

When set to Display anonymous, the modem hides your caller ID. When set to All anonymous, the modem

hides both caller ID and the SIP URL of the originating call.

DND (Do Not Disturb)

Select this option if your VoIP Service Provider has enabled DND (Do Not Disturb) on your SIP account and you

wish to use this feature.

Enable T38 Redundancy

Support

Select this function if you wish to send or receive faxes via VoIP and have a fax machine capable of using the

T38 fax over VoIP protocol.

Enable VBD redundancy

support

Select this checkbox to use the feature.

Enable VAD support

Enables the Voice Activated Detection function of the modem.

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Enable RTCP Flow Control

Select this checkbox to use the feature.

Enable Echo Cancellation

Select this checkbox to use the feature.

Enable # To ASCII

Select this checkbox to use the feature.

Enable Reinjection Function

Select this checkbox to use the feature.

RFC2198 Payload Value

(range 97-127)

Enter the RFC2198 payload value that the valid range is 96 ~ 127.

Registration Expire Timeout

Enter the registration expire timeout.

Session Expire Time

The interval of dialog refreshing time.

Min Session Expire Time

The minimum interval of dialog refreshing time.

VoIP DialPlan Setting

Set the VoIP dial plan. If user-dialed number matches it, the number is processed by the DSL router

immediately.

DSCP for SIP

Set the DSCP for SIP. You can select it from the drop-down list.

DSCP for RTP

Set the DSCP for RTP. You can select it from the drop-down list.

Dtmf Relay Setting

Set DTMF transmit method, which can be following values:

–

SIP Info: Use SIP INFO message to transmit DTMF digits.

–

RFC2833: Use RTP packet to encapsulate DTMF events, as specified in RFC 2833.

–

InBand: DTMF events are mixed with user voice in RTP packet.

SIP Transport Protocol

Select the transport protocol to use for SIP signaling. Note that the SIP proxy and registrar need to support the

protocol you select.

Enable Local Supplementary

Service

Select the checkbox to enable the supplementary service settings by the telephone set. If you deselect the

checkbox, the supplementary service can not be set by the telephone set.

Table 4: VoIP - Advanced - Service Provider

SIP Extra Setting

This page displays additional settings related to the SIP service.

PARAMETER

DEFINITION

Dial tone time

Dial tone duration.

Busy tone time

Busy tone duration.

Inter digit time

The valid range is 1 ~ 5.

Offhook warning tone time

Offhook warning tone duration.

Ringback tone time

Ringback tone duration.

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SIP Debug Setting

This page allows you to configure various settings regarding the logging levels of the SIP service.

OPTION

DEFINITION

SIP Log Server IP Address

Enter the IP address where the SIP Log data for the router’s currently saved VoIP account settings will be sent

to.

SIP Log Server port

Enter the port to be used for transmitting the SIP Log data for the router’s currently saved VoIP account

settings.

Ingress Gain

The incoming signal amplitude can be controlled with this field. Combined with the Egress gain a ratio can be

expressed of input to output. The Ingress Gain setting can help improve the quality of the VoIP line, and can

influence call volumes and help eliminate echoes.

Egress Gain

The outgoing signal amplitude can be controlled with this field. Combined with the Ingress gain a ratio can be

expressed of input to output. The Egress Gain setting can help improve the quality of the VoIP line, and can

influence call volumes and help eliminate echoes.

VoIP Functionality

This section describes how to use the VoIP function of the DSL router in more detail. Some features involve 2 or 3 parties. In that

case, note that all 3 parties have to be successfully registered.

Registering

Before using any VoIP functions, the DSL router has to register itself to a registrar. The DSL router also has to be configured with

a proxy, which relays VoIP signaling to the next hop. In fact, many implementations integrate these two into one server, so in

many case registrar and proxy refer to the same IP.

1.

Select the right interface to use for registering, depending on where proxy/registrar resides. If use WAN link, ensure that

it is already up.

2.

Select the checkbox of Use SIP Registrar, and fill in the IP address and port with the right value.

3.

Fill the extension information: Authentication name, Password, Cid Name and Cid Number.

4.

Click Apply to take the settings into effect.

5.

TEL indicator of VoIP service should be on, indicating that SIP client is successfully registered.

Placing a Call

This section describes how to place a basic VoIP call.

1.

Pick up the receiver on the phone.

2.

Hear the dial-tone. Dial the extension of remote party.

3.

To end the dialing, wait for digit timeout, or just press # immediately.

4.

After the remote party answers the call, you are in voice connection.

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Anonymous call

Anonymous call does not send the caller ID to the remote party. This is useful if you do not want others know whom you are.

1.

Enable Anonymous calling in the Voice--SIP Advanced Setting web page.

2.

Pick up the receiver on the phone.

3.

Dial *68 to enable anonymous call.

4.

Hook on the receiver, and dial another extension as you like. Now your caller ID information is blocked.

Do Not Disturb (DND)

If DND is enabled, all incoming calls are rejected. DND is useful if you do not want others to disturb you.

1.

Enable DND in the Voice--SIP Advanced Setting web page.

2.

Pick up the receiver on the phone.

3.

Dial *78 to enable DND.

4.

Hook on the phone. Now your phone rejects all incoming calls.

5.

Hook off again to disable the DND.

Call Return

For incoming calls, the DSL router remembers the number of calling party.

1.

Enable Call Return in the Voice--SIP Advanced Setting web page.

2.

Press *69 to return a call.

3.

Now you can make the call as if you have dialed the whole number.

Call Hold

Call hold enable you to put a call to a pending state, and pick it up in future.

1.

Assuming you are in a voice connection, you can press FLASH to hold current call.

2.

Now you can call another party, or press FLASH again to return to first call.

Call Waiting

Call waiting allows third party to call in when you are in a voice connection.

1.

Enable Call waiting in the Voice--SIP Advanced Setting web page.

2.

Pick up the phone attached to the DSL router.

3.

Assuming you are in a voice connection. When another call comes in, the DSL router streams a call waiting tone to your

phone, indicating another call is available.

4.

Press FLASH to switch to this call and the initial call put to hold automatically.

5.

Press FLASH multi-times to switch between these two calls back and forth.

Blind Transfer

Blind transfer transfers the current call to a third party blindly, regardless of whether the transfer is successfully or not.

1.

Assume you have already been in a voice connection.

2.

Press FLASH to hold the first party.

3.

Dial #90 + third party number.

4.

Before the third party answering the call, hook on your phone.

5.

Now the first party takes over the call and he is in connection with the third party.

Consultative Transfer

Consultative transfer lets the third party answer the transferred call, and then hook on the transferring party. It

’

more gentle than

blind transfer.

1.

Assume you have already been in a voice connection with a first party.

2.

Press FLASH to hold the first party.

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3.

Dial #91 + third party number.

4.

After the third party answering the call, hook on your phone.

5.

Now the first party takes over the call and he is in connection with the third party.

Call Forwarding No Answer

If this feature enabled, incoming calls are forwarded to third party when you does answer them. It involves in two steps: setting the

forwarding number and enable the feature.

1.

Enable Forward on "no answer" in the Voice--SIP Advanced Setting web page.

2.

When our phone does not answer the incoming call, the call is forwarded.

Call Forwarding Busy

If this feature enabled, incoming calls will be forwarded to third party when you busy. It involves two steps: setting the forwarding

number and enable the feature.

1.

Set Busy Call forwarding number and enable Forward on "busy" in the Voice--SIP Advanced Setting web page.

2.

When our phone is busy, this call can be forwarded.

Call Forwarding All

If this feature enabled, incoming calls are forwarded to third party without any reason. It involves in two steps: setting the

forwarding number and enable the feature.

1.

Set Unconditionally Call forwarding number and Forward unconditionally in the Voice--SIP Advanced Setting web page.

2.

All incoming calls are forwarded to the third party.

Three-Way Conference

Three-way conference enables you to invite a third party to a call, and every person in the conference is able to hear others

’

voice.

1.

Assume you are in connection with a first party.

2.

Press FLASH to put the first party on-hold.

3.

Dial a third party.

4.

After the third party answers the call, press FLASH again to invite the first party.

5.

Now all three parties are in a three-way conference.

T38 Faxing

To make T38 faxing, enable T38 support on the Web. After that, connect a fax machine to a FXS port of the DSL router. Now you

can use it as a normal phone, and it is able to send or receive fax to or from other fax machines on the VoIP network.

In the initial setup, faxing behaves like a normal call. After the DSL router detects the fax tone, it switch to T38 mode, and use it as

the transmit approach.

Pass-Through Faxing

If T38 support is disabled, faxing uses normal voice codec as its coding approach. Therefore, this mode more like normal phone

calls.