Linksys SPA-9000 Router Manual PDF (Setup & Configuration Guide)

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Chapter 6: Using the Web-based Utility

The Voice Tab

IP Telephony System

is *72. After the user dials *72, the System will wait for the user to enter a phone number. After the number has

been entered, the System will forward all calls for that phone number.

Vertical Service Announcement Codes

Service Annc Base Number

. Enter the base number for service announcements.

Service Annc Extension Codes

. Enter the extension codes for service announcements.

Outbound Call Codec Selection Codes

Prefer G711u Code

. This is the dialing code that will make this codec the preferred codec for the associated call.

The default is

*017110

.

Force G711u Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*027110

.

Prefer G711a Code

. This is the dialing code that will make this codec the preferred codec for the associated call.

The default is

*017111

.

Force G711a Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*027111

.

Prefer G723 Code

. This is the dialing code that will make this codec the preferred codec for the associated call.

The default is

*01723

.

Force G723 Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*02723

.

Prefer G726r16 Code

. This is the dialing code that will make this codec the preferred codec for the associated

call. The default is

*0172616

.

Force G726r16 Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*0272616

.

Prefer G726r24 Code

. This is the dialing code that will make this codec the preferred codec for the associated

call. The default is

*0172624

.

Force G726r24 Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*0272624

.

Figure 6-33: Voice - Regional Screen - Vertical Service

Announcement Codes

Figure 6-34: Voice - Regional Screen - Outbound Call

Codec Selection Codes

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Chapter 6: Using the Web-based Utility

The Voice Tab

IP Telephony System

Prefer G726r32 Code

. This is the dialing code that will make this codec the preferred codec for the associated

call. The default is

*0172632

.

Force G726r32 Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*0272632

.

Prefer G726r40 Code

. This is the dialing code that will make this codec the preferred codec for the associated

call. The default is

*0172640

.

Force G726r40 Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*0272640

.

Prefer G729a Code

. This is the dialing code that will make this codec the preferred codec for the associated call.

The default is

*01729

.

Force G729a Code

. This is the dialing code that will make this codec the only codec that can be used for the

associated call. The default is

*02729

.

Miscellaneous

Set Local Date (mm/dd)

. Set the local date (mm stands for months and dd stands for days). The year is optional

and uses two or four digits.

Set Local Time (hh/mm)

. Set the local time (hh stands for hours and mm stands for minutes). Seconds are

optional.

Time Zone

. For caller ID generation, select the number of hours to add to GMT to generate the local time. The

default is

GMT-08:00

.

FXS Port Impedance

. This sets the electrical impedance of the FXS port. Select one of these choices:

600

,

900

,

600+2.16uF

,

900+2.16uF

,

270+750||150nF

,

220+850||120nF

,

220+820||115nF

, or

370+620||310nF

. The

default is

600

.

Daylight Saving Time Rule

. Enter the rule for calculating daylight saving time; it should include the start, end,

and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below.

Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by

0:0:0 of the given date.

This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>

The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each

value is in this format: <month> /<day> / <weekday>[/HH:[mm[:ss]]]

Figure 6-35: Voice - Regional Screen - Miscellaneous

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56

Chapter 6: Using the Web-based Utility

The Voice Tab

IP Telephony System

The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during

daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired

instead of addition. The <save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]

The <month> value equals any value in the range 1-12 (January-December).

The <day> value equals [+|-] any value in the range 1-31.

If <day> is 1, it means the <weekday> on or before the end of the month (in other words the last occurrence of <

weekday> in that month).

The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0.

If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In

that case, the <day> value must not be negative.

If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the

<weekday> value on or after the date given.

If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the

<weekday> value on or before the date given.

The abbreviation HH stands for hours (0-23).

The abbreviation mm stands for minutes (0-59).

The abbreviation ss stands for seconds (0-59).

The default Daylight Saving Time Rule is

start=4/1/7;end=10/-1/7;save=1

.

FXS Port Input Gain

. Enter the input gain in dB, up to three decimal places. The range is 6.0 to -infinity. The

default is

-3

.

FXS Port Output Gain

. Enter the output gain in dB, up to three decimal places. The range is 6.0 to -infinity. The

default is

-3

.

DTMF Playback Level

. Enter the local DTMF playback level in dBm, up to one decimal place. The default is

-16

.

DTMF Playback Length

. Enter the local DTMF playback duration in milliseconds. The default is

.1

.

Detect ABCD

. To enable local detection of DTMF ABCD, select

yes

. Otherwise, select

no

. The default is

yes

.

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Chapter 6: Using the Web-based Utility

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IP Telephony System

Playback ABCD

. To enable local playback of OOB DTMF ABCD, select

yes

. Otherwise, select

no

. The default is

yes

.

Caller ID Method

. You have a choice of methods to use for caller ID. Select

Bellcore(N.Amer, China)

for CID,

CIDCW, and VMWI. FSK is sent after the first ring, and there is no polarity reversal or DTAS. Select

DTMF(Finland,Sweden)

for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring.

Select

DTMF(Denmark)

for CID only. DTMF is sent after polarity reversal (with no DTAS) and before the first ring.

Select

ETSI DTMF

for CID only. DTMF is sent after DTAS (with no polarity reversal) and before the first ring. Select

ETSI DTMF With PR

for CID only. DTMF is sent after polarity reversal and DTAS and before the first ring. Select

ETSI DTMF After Ring

for CID only. DTMF is sent after the first ring (with no polarity reversal or DTAS). Select

ETSI FSK

for CID, CIDCW, and VMWI. FSK is sent after DTAS (with no polarity reversal) and before the first ring. It

will wait for ACK from CPE after DTAS for CIDCW. Select

ETSI FSK With PR(UK)

for CID, CIDCW, and VMWI. FSK is

sent after polarity reversal and DTAS and before the first ring. It will wait for ACK from CPE after DTAS for CIDCW.

Polarity reversal is applied only if the equipment is on-hook. The default is

Bellcore(N.Amer, China)

.

Caller ID FSK Standard

. The System supports bell 202 and v.23 standards for caller ID generation. Select the

FSK standard you want to use,

bell 202

or

v.23

. The default is

bell 202

.

Feature Invocation Method

. Select the method you want to use,

Default

or

Sweden default

. The default is

Default

.

When you have finished making changes, click the

Submit All Changes

button to save the changes, or click the

Undo All Changes

button to undo your changes.

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Chapter 6: Using the Web-based Utility

The Voice Tab

IP Telephony System

The Voice - FXS 1/2 Screen

Use the appropriate screen to configure settings for each FXS port, which is called the Phone port on the System.

Line Enable

. To enable this line for service, select

yes

. Otherwise, select

no

. The default is

yes

.

Network Settings

SIP ToS/DiffServ Value

. Enter the TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default

is

0x68

.

SIP CoS Value

. Enter the CoS value for SIP messages. The default is

3

.

RTP ToS/DiffServ Value

. Enter the ToS/DiffServ field value in UDP IP packets carrying RTP data. The default is

0xb8

.

RTP CoS Value

. Enter the CoS value for RTP data. The default is

6

.

Network Jitter Level

. This setting determines how jitter buffer size is adjusted by the System. Jitter buffer size

is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP

frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger

for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the

minimum. Select the appropriate setting:

low

,

medium

,

high

,

very high

, or

extremely high

. The default is

high

.

Jitter Buffer Adjustment

. This controls how the jitter buffer should be adjusted. Select the appropriate setting:

up and down

,

up only

,

down only

, or

disable

. The default is

up and down

.

SIP Settings

SIP Port

. Enter the port number of the SIP message listening and transmission port. The default is

5080

.

SIP Remote-Party-ID

. To use the Remote-Party-ID header instead of the From header, select

yes

. Otherwise,

select

no

. The default is

yes

.

SIP Debug Option

. SIP messages are received at or sent from the proxy listen port. This feature controls which

SIP messages to log. Select

none

for no logging. Select

1-line

to log the start-line only for all messages. Select

1-line excl. OPT

to log the start-line only for all messages except OPTIONS requests/responses. Select

1-line

IMPORTANT:

In most cases, you should not change the service settings unless instructed to do

by your ITSP.

Figure 6-36: Voice - FXS 1 Screen - Network Settings

Figure 6-37: Voice - FXS 1 Screen - SIP Settings

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