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WiFi Data and VoIP Gateway

NF1ADV

OPTION

DEFINITION

PSTN SETTINGS

Incoming PSTN Call Routing

There are two options for how PSTN or non-VoIP calls will be routed. They are:



Auto

–

PSTN Call Switch to Idle Line. The PSTN call will be directed to any available handset.



Line - PSTN Call Switch to Physical Line. A small combo box appears for the router administrator

to select a handset to receive all PSTN phone calls.

PSTN Dial Plan For Outgoing

Calls

This field numbers can be entered as a prefix that will be automatically dialed before the user dials a phone

number when using the PSTN line to make a phone call. To use the PSTN line to make a phone call dial ## and

then the phone number.

VoIP SETTINGS

SIP Account

This field gives the option of selecting a single VoIP account configuration or multiple VoIP account

configurations. Please note multiple VoIP accounts can only be configured using one VoIP Service Provider.

Bound interface Name

Select the Interface that the VoIP account will use to make a connection to the VoIP Service Provider.

SIP SETTINGS

Locale Selection

The locale selection establishes the type of codec and the dial and ring tones for a given country.

SIP Domain Name

Enter the SIP domain name or IP address of your VoIP Service Provider here.

Max Digits Setting

Enter the maximum number of digits that a phone number can have.

Use SIP Proxy

Select this option if required by your VoIP Service Provider. Enter the SIP Proxy Domain Name and SIP Proxy

Port which is typically 5060.

Use SIP Outbound Proxy

Select this option if required by your VoIP Service Provider. Enter the SIP Proxy Domain Name and SIP Proxy

Port which is typically 5060.

Use SIP Registrar

Select this option if required by your VoIP Service Provider. Enter the SIP Proxy Domain Name and SIP Proxy

Port which is typically 5060.

Account Enabled

Use this option to enable or disable the VoIP account.

VoIP Phone Number

Enter the VoIP phone number as supplied to you by your VoIP Service Provider.

Display Name

Enter the Display Name as supplied to you by your VoIP Service Provider. This can be your VoIP Phone

Number.

Auth ID

Enter the Authorisation ID as supplied to you by your VoIP Service Provider.

Auth Password

Enter the Authorisation Password as supplied to you by your VoIP Service Provider.

Preferred ptime

The 'Preferred ptime' is the time delay in milliseconds between voice packets sent. You may wish to change

this setting depending on the account performance.

Preferred Codec 1

–

6

A codec is a method of compressing speech. More compression requires less

bandwidth but can sound worse. Typically, a phone will have a preferred codec, but will work with others. Use

the codec recommended by your service provider.

Table 13: Advanced

–

VoIP

–

SIP Basic Settings

After entering your VoIP settings press the Apply button. Select Management > Save/Reboot and press the Reboot button. Once

the router restarts if there is a valid internet connection and the VoIP account settings are valid the VoIP service will start.

To check if the VoIP service is working check your phone handset for a dial tone or navigate to Basic > Home and check that the

DECT and Phone registration status is displaying “Up” in the router web interface.

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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 24: VoIP - Advanced - Service Provider

OPTION

DEFINITION

Call Waiting

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

Call Forwarding Number

Enter the phone number to be forwarded to if your VoIP Service Provider has enabled Call Waiting on your SIP

account and you wish to use this feature.

Forward Unconditionally

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 “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.

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 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.

Interdigit Timeout

The time in seconds before which a number must be dialed or become an invalid number.

Registration Expire Timeout

The time in minutes for the SIP registered will be renewed.

Registration Retry Interval

The time in minutes before the SIP settings will attempt to be registered.

DSCP for SIP

DSCP (Differentiated Services Code Point) for SIP (Session Initiation Protocol) relates to QoS (Quality of Service)

settings. Only use this field if directed by your network administrator.

DSCP for RTP

DSCP (Differentiated Services Code Point) for RTP (Real Time Protocol) relates to QoS (Quality of Service)

settings. Only use this field if directed by your network administrator.

DTMF Relay Settings

Dual-tone Multi-frequency Relay (DTMF) is the mechanism whereby a local Voice over IP (VOIP) gateway listens

for DTMF digits (during a call), and then sends them uncompressed as either RTP or H.245 packets to the

remote VOIP gateway, which regenerates DTMF digits and prevents digit loss due to compression.

RTP Payload Type for

RFC2833

The Real Time Protocol Payload type for RFC2833. RFC2833 is a standards-based mechanism used to send

DTMF digits in-band (RTP) that is supported by many vendors in the industry.

Hook Flash Relay Setting

A hookflash is a brief interruption in the loop current on loopstart trunks that the attached system

does not interpret as a call disconnect. Once the PBX or PSTN senses the hookflash, it generally puts the

current call on hold and provides a secondary dial tone or access to other features such as transfer or call

waiting access.

A hookflash is done by momentarily pressing down the cradle on a telephone. Some telephone handsets have

a button called 'flash' or 'recall' that sends a 'timed loop break', or 'calibrated flash' which is a hookflash that

has a precise timing.

SIP Transport Protocol

The protocol used to transport SIP traffic. This is almost always UDP.

Enable SIP Tag Matching

Select this option to enable SIP Tag Matching.

Table 14: VoIP - Advanced - Service Provider

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SIP Debug Settings

This page allows you to set the IP address where the SIP Log data for the router VoIP account settings will be sent to and the port

number through which it will be sent.

Figure 25: VoIP - Debug Settings

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.

VoIP Module Console Log

Level

Select the type of debug messages you would like to receive. The options are:



Error. Only error messages will be logged.



Notice. Only Notice messages will be logged.



Debug. All messages will be logged.

VAD Support

Select to enable Voice Activated Dialing for a given phone handset.

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.

Table 15: VoIP - Debug Settings

Adjusting Call Quality with the Ingress/Egress Gain Settings

If your call quality is poor with heavy echo and lag times try setting the Ingress Gain value to less than 0. With less ingress the sound

volume will be lower but should reduce line echo. The optimum quality to try to attain is clarity of audio signal both incoming and

outgoing, with good call volume and little perceived echo or distortion. However the values to use will vary and are dependent on

network bandwidth, associated hardware and software codecs used.



Carry out test call trials starting with both the Ingress and Egress Gain set to about

–

10. Values of -1 to -11 should provide

a clear audio stream with low echo and distortion.



Continue to lower the value one setting at a time, using increments of two or three.



Make test calls until the echo is moderated.

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DECT

The NF1ADV DECT settings page displays status information and allows for DECT cordless p

hones to be registered to the router’s

on-board DECT base station. The NF1ADV can function as a DECT (Digital Enhanced Cordless Telecommunications) base station

for up to 4 cordless phones for both VoIP or PSTN calling.

Figure 26: VoIP

–

DECT

OPTION

DEFINITION

DECT

–

General Module Information

Module Identifier

The MAC address of the DECT base station.

Manufacturer Identifier

This is an 8 bit unique ID of the DECT base station in the form of an EMC (Equipment Manufacturer Code).

Model Identifier

This is an8 bit model ID that is unique for the DECT base station model and associated firmware version.

DECT

–

Base Station: Information and Action

DECT Interface Status

This field shows whether the DECT base station is enabled or disabled.

Maximum Number of

Handsets

This field shows the number of DECT cordless phones that can be connected to the DECT base station at one

time.

Currently Registered handsets

This field shows the number of DECT cordless phones that can are currently connected to the DECT base

station.

Registration Window

Use this field when registering a DECT cordless phone to the router.

Station Registration Access

Code

To set the access code, enter 4 numbers and press the Set Code button. Only DECT cordless phones that use

the correct access code can connect to the DECT base station and so use your network bandwidth. We

recommend not using the default “0000” value.

DECT

–

Handset: Information and Action

Handset Identifier

This field shows the DECT handset number as set in the DECT base station settings on handset was

registration.

Status

This field shows the current status of the DECT handset connection.

Subscription Time

This field shows the date and time that the DECT handset was connected.

International Portable

Equipment Identity (IPEI)

A 36 bit unique identifier of the DECT handset.

Manufacturer Identifier

This is an 8 bit unique ID of the DECT handset in the form of an EMC (Equipment Manufacturer Code).

Model Identifier

This is an8 bit model ID that is unique for the DECT handset model and associated firmware version.

Table 16: VoIP - DECT Settings

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Connecting a Cordless Phone to the DECT Base Station

1.

In the NF1ADV web interface select VOICE > DECT.

2.

Set a 4 digit Station Registration Access Code and press the Set Code button.

3.

On your DECT cordless phone navigate to the Base Registration Setting in the Advanced Settings.

4.

If prompted set the DECT phone to handset “X” where “X” is the number of DECT handsets + 1 that are alr

eady connected to

the NF1ADV DECT base station.

5.

Press the DECT button on the router for 5 seconds or press the Start Registration button in the DECT page of the NF1ADV

web interface.

6.

If the router detects the phone correctly you should now be prompted for a registration pin on the DECT handset. Enter the

Station registration Access Code you set in Step 2 into the Cordless Phone and press Ok or Apply.

7.

The cordless phone

will give recognition that it is connected to the router’s DECT base station in the form

of an audio beep or

test message in the handset’s interface.

8.

Press the Ping All Handsets button to verify the handset is connected to the base station. The DECT phone should produce a

series of audio sounds if the DECT phone is still communicating with the NF1ADV DECT base station.