Alvarion BMAX-6000 Telrad Router Manual PDF (Setup & Configuration Guide)

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BreezeMAX PRO 6000 Product Manual

Chapter 9 - Engineering

VoIP Configuration

89

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.

N/A

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

Customized

Caller ID



Select the standard by which the caller

is identified:



British TelecomDual-tone

multi-frequency signaling standard

US

G711 Fax Codec

Select the codec to convert fax signals into

digital data to be transmitted over the

Internet.

g711a

g711u

g711a

Modem Call Codec

Select the codec to be used for modem

calls; when a modem call is detected, this

codec will be used

g711u

g711a

Page 112 / 138

BreezeMAX PRO 6000 Product Manual

Chapter 9 - Engineering

VoIP Configuration

90

Hook Flash

Max/Min. Timer

Enter a value (in milliseconds) to define

how long should the hook be pressed as to

be considered as flash (hook should be

pressed for a time between min. and max.

values)

Max: 0.9 ms

Min: 0.1 ms

100 - 1550 ms

Registration Expire

Enter a value (in seconds) to define the

time by which the CPE has to renew its

subscription to the SIP server

3600 seconds

1 - 99999 seconds

Enable Telmex FQDN

Enable Request for Comments (RFC) 3263

"Locating SIP Servers" functionality.

The

Session Initiation Protocol (SIP) uses DNS

procedures to allow a client to resolve a SIP

Uniform Resource Identifier (URI) into the

IP address, port, and transport protocol of

the next hop to contact. It also uses DNS

to allow a server to send a response to a

backup client if the primary client has

failed. This procedure uses the

Fully-qualified domain name (FQDN).

Uncheck

Check/Uncheck

Customized Tone Settings

Default Dial Tone

Defines the tone that will be heard during

dialing. The string refers to tone,

frequency, and cadence.

350@-19,440@-19;1

0(*/0/1+2)

Set by the operator

Default Callwaiting

Tone

Defines the tone that will be heard during

call waiting. The string refers to tone,

frequency, and cadence.

440@-22;31.2(.3/10.1

/1)

Default MWI Tone

Defines a message-waiting indicator tone.

The string refers to tone, frequency, and

cadence.

350@-13,440@-13;1.

2(.1/.1/1+2);*(*/0/1+

2)

Known SIP Provider

Enable WiMAX QoS

For known SIP

Provider

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.

Enabled

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

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

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BreezeMAX PRO 6000 Product Manual

Chapter 9 - Engineering

VoIP Configuration

91

Line 1 Settings - Common Settings

Enable Line 1

To enable voice feature

Disabled

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.

Disabled

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.

Disabled

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.

Disabled

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

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.

Enable

Enable/disable

Anonymous Call

Reject

Select this check-box to block calls from an

unidentified number.

Disabled

Enable/disable

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

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BreezeMAX PRO 6000 Product Manual

Chapter 9 - Engineering

VoIP Configuration

92

E911

Emergency call: Enter a number that will

be referred to as the emergency call. When

dialing “911” this call will be routed to the

emergency service.

N/A

Automatic Recall

Return call: Enables calling back the

number whose call you missed. Enter a

special number (e.g. *42). Dialing this

number will recall the number that was

missed in last incoming call. Empty field

means Automatic Return Call is disabled

N/A

Redial

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

redialing to the last number

N/A

Automatic Call Back

Repeat dial if busy: automatically redial the

number time and again. Define a special

number (e.g.*52). Dialing this number

after the busy tone received, will

automatically redial the number until the

recipient's line is free. Then your phone

will ring back when you are being

connected. Empty field means Automatic

Return Call is disabled.

N/A

Inter-digit T/O

Delay in call establishment (timeout in

seconds)

5 sec.

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.

N/A

DialPlan

Establish the expected number and pattern

of digits for a telephone number

N/A

Flash Timeout

When pressing Flash you have the time

interval defined by this value to dial other

numbers (e.g. for a conference call). If you

do not dial a number within the specified

time, you return to the initial call.

15 seconds

Any number

Date Mode

Use date information from the Date

header in the SIP message/NTP server

Date Header

Date Header/NTP

Enable Hold Tone

Select whether to play hold tone when put

in hold.

Check

Check/Uncheck

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

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BreezeMAX PRO 6000 Product Manual

Chapter 9 - Engineering

VoIP Configuration

93

CLIR per-call

Enter a dialing prefix for Calling Line

Identification Restriction (CLIR) per call, for

example: *45

N/A

Any number

CallHold

Enables holding the line while speaking

with one participant in a conversation.

Disabled

Enable/disable

Do Not

Disturb(DND)

Select this check-box to reject any

incoming calls. The call will result in Busy

tone.

Disabled

Enable/disable

Codec Setting

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

Third priority

g711u ptime

Set the time (in milliseconds) 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.

Enabled

Enable/disable

g711a Priority

The priority of codec by which the unit will

attempt to use for best voice quality

Second priority

g711a ptime

Set the time (in milliseconds) 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.

Enabled

Enable/disable

Table 9-2: VoIP Settings (Continued)

Parameter

Description

Default

Possible Values

Rate

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