Alvarion WIXFBR-103X187 Router Manual PDF (Setup & Configuration Guide)

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Chapter 9 - Engineering

VoIP Configuration

BreezeMAX Si 4000 CPE

112

Operator Manual

Figure 9-4: Engineering - VoIP (continued)

Page 127 / 149

Chapter 9 - Engineering

VoIP Configuration

BreezeMAX Si 4000 CPE

113

Operator Manual

The following table describes the VoIP Settings parameters:

Table 9-2: VoIP Settings

Parameter

Description

Default

Possible Values

Global Settings

user Domain

The host portion of the SIP Uniform Resource

Identifiers (URIs) that are assigned to users in

a network. The SIP domain name can

sometimes be different from the internal

network domain name.

N/A

Up to 256 characters

registrar Address

The IP address of the SIP registrar server. A

registrar is a server that accepts SIP register

requests and places the information it

receives in those requests into the location

service for the domain it handles.

N/A

Up to 256 characters

registrar Port

The TCP port number used by the VoIP

service provider’s register server.

5060

Range: 1030 to 65535

outbound Proxy

Address

Address of the VoIP service provider SIP

proxy server.

Up to 256 characters

outbound Proxy Port

The TCP port number used by the VoIP

service provider’s SIP proxy server.

5060

Range: 1030 to 65535

RTP Port Range Start

Enter the port Start and End to define the

range that Real-time Transport Protocol will

use

8000

Range: 1030 to 65535

RTP Port Range End

8015

DSP Nation

National protocol definition

Default

Caller ID

Select the standard by which the caller is

identified:



Bellcore - (Bell Communications

Research) - used in the USA, Canada,

Australia, China, Hong Kong and

Singapore



ETSI FSK - European

Telecommunications Standards Institute

Frequency Shift Keying

-



V-23 FSK - developed by NTT in Japan



BT=>4 - British Telecom standard

Bellcore

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Chapter 9 - Engineering

VoIP Configuration

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Operator Manual

Known SIP Provider

Enable WiMAX QoS

For known SIP

Provider

(Not available for this CPE) This check-box

enables the CPE to select the quality of

service level from a known SIP. This feature is

used for MS initial service -flow.

N/A

Enable/disable

List of SIP providers

Click

Insert

to add a known SIP provider to

the list and specify the SIP Proxy address and

Proxy port. To remove from the list, click

Del

.

N/A

Line 1 Settings - Common Settings

Enable Line 1

To enable voice feature

Enable/disable

DTMF Method

Enable the sending of dual-tone

multi-frequency (touch tone) phone signals

over the VoIP connection:



InBand - The DTMF signals are sent over

the RTP voice stream.



RFC2833 - Relay the DTMF signals over

the RTP voice stream without any

distortion



RFC2833+InBand - Uses the best method

depending on the called party.



SIPInfo - Uses the data from SIP

RFC2833+

InBand

InBand, RFC2833,

RFC2833+InBand,

SIPInfo

callForward

Unconditional

Forwards an incoming call to another number

for all calls.

Disable

Enable/disable

callForward

Unconditional Number

Enter the number to which to forward all

incoming calls.

N/A

Up to 256 characters

callForward

Busy

Forwards an incoming call to another number

when the current line is busy.

Disable

Enable/disable

callForward

Busy Number

Enter the number to which to forward

incoming calls when the current line is busy.

N/A

callForward

NoReply

Incoming calls are forwarded to another

phone number only if there is no answer after

a pre-configured timeout.

Disable

Enable/disable

Call Forwarding No

Reply Timeout

The time (in seconds) a call waits for an

answer before being forwarded to the number

specified in callForward NoReply

30 seconds

N/A

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

Page 129 / 149

Chapter 9 - Engineering

VoIP Configuration

BreezeMAX Si 4000 CPE

115

Operator Manual

callForward

NoReply Number

Enter the number to which to forward

incoming calls when there is no reply from

current line.

N/A

Caller ID Block

Select this check-box to hide your name and

number when calling another number.

Disabled

Enable/disable

E911

Emergency call: Enter a number that will be

referref as the emergency call. When dialing

“911” this call will be routed to the emergency

service.

N/A

Redial

Enter a shortcut (e.g. *53) to define redialing

to the last number

Inter-digit T/O

timeout in seconds

call Waiting T/O

timeout in seconds

DialPlan

Establish the expected number and pattern of

digits for a telephone number

CallHold

Enables holding the line while speaking with

one participant in a conversation .

Enable/disable

call Waiting

Enables suspending the current telephone

call and switch to a new incoming call.

Enable/disable

Do Not Disturb(DND)

Select this checkbox to reject any incoming

calls. The call will result in Busy tone.

Disable

Enable/disable

Anonymous Call

Reject

Select this checkbox to block calls from an

unidentified number.

Disable

Enable/disable

Automatic Recall

Return call: Enables calling back the number

whose call you missed by pressing some

buttons.

Enable/disable

Automatic Call Back

Repeat dial if busy: automatically redial the

number time and again, until the recipient's

line is free. Then your phone will ring back

when you are being connected.

Enable/disable

Call Switching

Set a shortcut (e.g. *66) to enable

switching

from one phone to another without hanging

up. Switching is done by pressing the flash

button and dialing the shortcut number.

Codec Setting

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

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Chapter 9 - Engineering

VoIP Configuration

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Operator Manual

g711u Codec Enable

The ITU-T G.711 with mu-law standard codec

that uses Pulse Code Modulation (PCM) to

produce a 64 Kbps high-quality voice data

stream. This standard is used in North

America and Japan.

Enable

Enable/disable

g711u Priority

The priority of codec by which the unit will

attempt to use for best voice quality

Fourth

priority

g711u ptime

Set the time (in miliseconds ) for the unit to

attempt to use the codec highest priority in the

list before trying the next lower one.

30 ms

g711a Codec Enable

(G711.aLaw): The ITU-T G.711 with A-law

standard codec that uses Pulse Code

Modulation (PCM) to produce a 64 Kbps

high-quality voice data stream. This standard

is used in Europe and most other countries

around the world.

Enable

Enable/disable

g711a Priority

The priority of codec by which the unit will

attempt to use for best voice quality

Third

priority

g711a ptime

Set the time (in miliseconds ) for the unit to

attempt to use the codec highest priority in the

list before trying the next lower one.

30 ms

g729 Codec Enable

The ITU-T G.729ab standard codec that uses

Conjugate Structure Algebraic-Code Excited

Linear Prediction (CS-ACELP) with silence

suppression to produce a low-bandwidth data

stream of 8 Kbps. Note that DTMF and fax

tones do not transport reliably with this codec,

it is better to use G.711 for these signals.

Enable

Enable/disable

g729 Priority

The priority of codec by which the unit will

attempt to use for best voice quality

First priority

g729 ptime

Set the time (in miliseconds ) for the unit to

attempt to use the codec highest priority in the

list before trying the next lower one.

30 ms

ILBC Codec Enable

Internet Low Bitrate Codec

Enable

Enable/disable

ILBC Priority:

The priority of codec by which the unit will

attempt to use for best voice quality

Last priority

ILBC ptime

Set the time (in miliseconds ) for the unit to

attempt to use the codec highest priority in the

list before trying the next lower one.

30 ms

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

Rate

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