ADSL2+ Wireless N300 Modem Router with VoIP User Guide

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Voice

This chapter first describes the various options for configuration of the SIP voice service.

It then provides detailed instructions for making

telephone calls using VoIP (Voice over IP) or PSTN (Public Switched Telephone Network) services

5.1 SIP

Session Initiation Protocol (SIP) is a peer-to-peer protocol used for Internet conferencing, telephony, events notification, presence and

instant messaging.

SIP is designed to address the functions of signalling and session management within a packet telephony network.

Signalling allows call

information to be carried across network boundaries.

Session management provides the ability to control the attributes of an end-to-end

call.

The SIP standard defines the following agents/servers:

1.

User Agents (UA) - SIP phone clients (hardware or software)

2.

Proxy Server – relays data between UA and external servers

3.

Registrar Server - a server that accepts register requests from UA

4.

Redirect Server – provides an address lookup service to UA

NOTE:

The SIP standard is set by the Internet Engineering Task Force (IETF).

The following subsections present Basic, Advanced and Debug SIP screens.

Each screen provides various options for customizing the SIP

configuration.

5.1.1

SIP BASIC

This screen contains basic SIP configuration settings.

Once settings are configured click

Save/Apply

to begin using the service.

NOTE:

Consult the tables that follow for detailed field descriptions

Interface name

Choose the WAN interface

Locale Selection

Sets tone, ring type and physical characteristics for specific countries

Preferred codecs

Choose G.711U, G.711A, G.726 or G.729

Preferred ptime

The time period used to digitally sample the analog voice signal.

The default is 20 ms.

Use SIP proxy

Enable the SIP proxy by selecting the checkbox and setting proxy parameters.

SIP Proxy

Input IP address or domain name of the SIP proxy server, used for VOIP service.

SIP Proxy port

This value is set by your VoIP provider and is normally port 5060.

Registration Expire Timeout

The time period the user would like the registration to be valid for the Registrar/ Proxy Server.

The

default is 300 seconds.

SIP domain name

Provided by your VoIP provider.

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ADSL2+ Wireless N300 Modem Router with VoIP User Guide

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A proxy is an intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other

clients.

Requests are serviced internally or transferred to other servers.

A proxy interprets and, if necessary, rewrites a request

message before forwarding it

Use SIP outbound

proxy

Select if required by your VoIP provider.

Enter SIP Outbound proxy IP and port.

Line 1 & 2

Ports FXS1 & FXS2

Disabled

Ticking the checkbox disables the line

Extension

The line extension number

Display Name

The caller ID display name

Authentication Name

The authentication username for the Registrar/Proxy, given by VOIP provider.

Authentication

Password

The authentication password for the Registrar/proxy, given by VOIP provider.

5.1.2

SIP ADVANCED

This screen contains advanced SIP configuration settings.

Once settings are configured click

Save/Apply

to begin using the service.

Line 1 & 2

Ports FXS1 & FXS2

Forwarding number

Enter the forwarding phone number

Call forwarding when

busy

Tick the checkbox to enable this option

Forwarding all calls

Tick the checkbox to enable this option

Call forwarding if no

answer

Tick the checkbox to enable this option

Call waiting (default:

enabled)

Tick the checkbox to enable this option

NOTE:

These options can also be set using telephone keypad commands, as described in the call command list of section 8.2 Telephone Calls

Enable MWI

Subscription

Enable or disable Message-Waiting Indicator (MWI) for FXS Phones with this checkbox.

Enable T.38 support

(default: enabled)

Enable or disable T.38 Fax mode support with this checkbox.

You can plug a fax machine into either phone port

to send or receive faxes.

Functionality depends upon FAX support by your VoIP service provider.

Max Digits

Sets the maximum number of digits for a phone number.

Emergency Setting

Multiple emergency numbers can be set using the “|” character (shift + backslash). For example, to set 911 and

114 as emergency numbers, enter “911|114”.

Please Note:

These numbers must be changed to correspond to the emergency numbers that are used in your

location.

Dtmf Relay Setting

Set the special use of RTP packets to transmit digit events.

SIP Transport

Protocol

Set the special use of SIP protocol to transmit digit events.

Incoming PSTN Call

Routing

If PSTN route rule is Auto, an incoming PSTN call will ring an idle phone, either Phone1 or Phone2 (if Phone1 is

busy).

If PSTN route rule is Line1 or Line2, an incoming PSTN call will attempt to ring only the assigned phone line

(FXS1 or FXS2).

Enable SIP tag matching

Select if required by your VoIP provider.

(e.g. disable with Vonage service.)

Enable Music Server

Enable/disable the Music Server.

Enter the Music Server IP address and port.

5.1.3

SIP DEBUG

ADSL2+ Wireless N300 Modem Router with VoIP User Guide

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This screen contains SIP configuration settings used for debugging.

Once settings are configured click

Save/Apply

to begin using the service.

5.2 Telephone Calls

To make a call, simply dial the number.

The dial plan (i.e. the dialled digits) is normally customized for each installation.

The default dial plan

allows for dialling of 4-digit extensions or direct IP addresses.

Shorter extension numbers (e.g. 3-digits) can be dialled by completing the

dial string with a final #.

When a Call Server (SIP Proxy Server) is configured into the system, the dialled digits are translated and routed by the Call Server to the

correct destination as registered with the Call Server.

If no Call Server is configured, calls can still be made using 4-digit extensions, rather than using full IP addresses.

The originator translates

the dialled-digits to a destination device as follows:

First Digit:

Line identifier (for multi-line gateways)

Remaining digits:

Host number part of an IP address.

The Network number part is considered to be the same

as the caller’s IP address.

For example, if a caller at address 10.136.64.33/24 dials “2023”, the call will be placed to the second line at address 10.136.64.23.

All

devices have to be on the same Class C subnet (24-bit subnet mask).

To dial an IP address directly, dial the IP address digits using * on the keypad as the dot.

Complete the address with a final * or #.

When

using IP address dialling it is not possible to specify which line at a gateway is called, so the gateway always routes IP-address dialled calls

to the first line.

Network busy tone (fast busy) will be played for unknown or unreachable destinations.

To answer a call, pick up the phone or press the

hands free button.

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ADSL2+ Wireless N300 Modem Router with VoIP User Guide

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NetComm Gateway

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CALL COMMAND LIST

Caller ID

The Call Manager delivers Caller ID when placing calls.

The caller ID is transmitted to the analog line for CLASS recognition.

Call Hold

To put a call on hold, press flash then hang up (optional).

To return to the original call, press flash or pick up the phone.

The phone will

issue a short ring burst every 30 seconds or so while on-hook to remind you that a call is on hold.

Call Transfer

•

To transfer a call, press flash then dial the new number.

•

To transfer immediately, hang up (blind transfer).

•

To transfer with consultation, wait for the party to answer, consult, and hang up.

Conference Calling

To turn a two-party call into a three-party conference call, press flash and dial the third party.

Wait for the party to answer, then press flash.

To drop the third party and return to a two-party call, press flash again.

To drop yourself out of the conference, hang up.

The call will be

transferred (so that the other two parties remain connected to each other).

NOTE:

In conference mode, the conference initiator performs the audio bridge/mixing function – there are only two voice streams established.

Call Waiting

If call waiting is enabled on a line, and you hear the call waiting tone during a call, press flash to answer the second call.

The first call is

automatically placed on hold.

To switch between calls, press flash again.

To disable the call waiting feature, dial *60.

To enable the call waiting feature, dial *61.

NOTE:

Call forward feature settings (Busy or All) take priority over the call-waiting feature.

The call-waiting feature is ignored on new incoming calls if there is already a call on

hold or in conference.

Call Forward Number

To set the call forward number, dial *74 then the number.

(Note that this does not actually enable forwarding; to do so, select the

call forward action as described below.)

To disable all call forwarding features, dial *70

Call Forward

No Answer

To enable call forward on no answer, dial *71. Incoming calls will be forwarded if unanswered for 18 seconds.

Call Forward Busy

To enable call forward if busy, dial *72. Incoming calls will forward immediately if the phone is off-hook.

Call Forward All

To enable call forward for all calls, dial *73.

To disable the “forward all calls” feature, dial *75. Settings for Call Forward Busy or

No Answer are not modified.

Call Return

To place a call to the last known incoming caller (unanswered or not), dial *69.

Redial

To redial the last outgoing number, dial *68.

Wireless