Cisco Systems IP Phone SPA9000 User Manual

Lin k s ys S PA9 0 0 0 Ad m in is t ra t o r Gu id e

Docum ent Version 3.01

Co rp o ra t e He a d q u a rt e rs

Linksys

121 Theory Drive

Irvine, CA 92617

USA

http://www.linksys.com

Tel: 949 823-1200

800 546-5797

Fax: 949 823-1100

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C O N T E N T S

Preface i-xi

D o cument A u dience i-xi

H o w T h is D o cument is O r ganized i-xii

D o cument C o nventions i-xii

R e lated D o cumentation i-xiii

T e chnical Support i-xiii

C H A P T E R

U s ing the L i nksys V o ice System 1-1

T h e L i nksys V o ice System 1-1

Ov e rview 1-1

SPA400 SIP-PSTN G a teway and V o icemail 1-2

A u to-A t tendant 1-3

SPA9000 System F e atures 1-3

A d ditional F e atures W hen U s ed with SP A 9 00 Series IP Phones 1-5

T e chnology B a ckground 1-6

Session Initiation Protocol 1-6

SPA9000 Media Proxy 1-7

U s ing the SP A 9 000 with a F i rewall or R o uter 1-8

SPA400 SIP-PSTN G a teway 1-8

N e twork A d dress T r anslation (N A T) 1-9

NAT Ov e rview 1-9

NAT Typ e s 1-10

Simple T r aversal of U DP T h rough N A T 1-10

SIP-NAT In t eroperation 1-11

SPA9000 Architecture 1-11

A r chitectural C o mponents 1-12

Multicast A d dressing and G r oup Paging 1-13

C o nfiguration O p tions 1-14

Interactive V o ice R e sponse 1-14

Setup Wizard 1-14

A d ministration Web Server 1-14

L o cal C l ient C o nfiguration and R e gistration 1-15

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Contents

R e m ote Provisioning of the SP A 9 000 1-15

Where to G o F r om H e re 1-16

C H A P T E R

G e tting Started 2-1

Implementing L VS 2-1

U s ing the SP A 9 000 and SP A 4 00 as a V o IP PBX Sy s tem 2-2

U s ing the SP A 9 000 as a Media Proxy 2-3

U s ing the L VS a s a K e y System 2-3

SPA9000 H a rdware 2-4

SPA9000 B a ck Panel 2-4

T h e F r ont Panel 2-5

SPA400 H a rdware 2-6

B a ndwidth R e quirements 2-6

C a ring for Y o ur H a rdware 2-7

M a k ing the Physical C o nnections 2-8

U s ing the Interactive V o ice R e sponse Interface 2-8

U s ing the I V R M e nu 2-9

I V R O p tions 2-10

E n tering a Password through the I V R 2-12

Initial Setup and C o nfiguration 2-12

L i censing 2-13

U s ing D HCP o r Static IP A d dressing 2-13

U s ing the Wizard for Initial C o nfiguration 2-13

U s ing the Wizard to U p grade Software 2-18

Setting the SP A 9 000 A d ministrator A c count Password 2-22

U s ing the A d ministration Web Server 2-22

C o nnecting to the A d ministration Web Server 2-22

A d ministrator A c count Privileges 2-24

A d vanced Methods of C o nfiguration 2-24

W e b I nterface U RLs 2-24

U p grade U RL 2-25

R e s y nc URL 2-25

R e boot U RL 2-25

Provisioning 2-26

Provisioning C a pabilities 2-26

C o nfiguration Profile 2-26

C l ient A u to-C o nfiguration 2-27

M a nual C l ient C o nfiguration 2-28

C l ient R e gistration 2-30

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T r oubleshooting and C o nfiguration F AQ 2-31

C H A P T E R

C o nfiguring V o ice Service and V o icemail 3-1

U s ing the Wizard to C o nfigure I T S P V o ice Services and V o icemail 3-1

C o mpleting the V o icemail C o nfiguration 3-6

C o nfiguring the SP A 4 00 for PS T N C o nnectivity or V o icemail 3-8

U n derstanding H o w the SP A 4 00 Interacts with the SP A 9 000 3-8

U s ing the SP A 9 000 Setup Wizard to C o nfigure the SP A 4 00 3-8

U s ing the A d ministration Web Server to C o nfigure the SP A 4 00 3-11

A c cessing the A d ministration Web Server 3-12

C o nfiguring the SP A 4 00 to C o nnect to the SP A 9 000 3-12

C o nfiguring the SP A 9 000 to R e gister the SP A 4 00 3-15

C o nfiguring the SP A 4 00 V o icemail Services 3-15

C o nfiguring the SP A 9 000 V o icemail Settings 3-16

C o nfiguring SP A 4 00 V o icemail A c counts 3-17

C o nfiguring a SPA IP Phone for V o ice Mail Service 3-18

SPA400 V o icemail O p tions 3-20

M a naging V o icemail 3-20

H o w V o icemail Works 3-21

C h ecking V o icemail from an E x t ernal Number 3-22

D e positing V o icemail 3-22

Subscribing to V o icemail Notification 3-23

C H A P T E R

C o nfiguring SP A 9000 F e atures 4-1

U s ing the Wizard to C o nfigure SP A 9 000 V o ice F e atures 4-1

C o nfiguring C l ient Stations 4-2

C o nfiguring C l ient E x t ensions 4-4

C o nfiguring Shared E x t ensions 4-5

C o nfiguring a C a ll H u nt G r oup 4-6

U s ing the Wizard for L o calization 4-7

U s ing D i al Plans 4-9

C o nfiguring D i al Plans 4-9

D i al Plan D i git Sequences 4-9

D i al Plan R u les 4-10

D i al Plan E x a m ples 4-11

D i al Plan T i mers 4-12

Interdigit L o ng T i mer 4-12

Interdigit Short T i mer 4-12

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D i al Plans 4-13

B a sic C a ll Management 4-13

R e ceiving E x t ernal P hone C a lls 4-13

C a lling B e tween C l ient Stations 4-13

C l ient Stations C a lling an E x t ernal Number 4-14

Ex t ernal U s ers C a lling the SP A 9 000 4-15

Supporting Multiple D ID Numb e rs Per L i ne Interface 4-16

M a naging C a ll F o rwarding 4-17

H o w C a ll F o rwarding Works 4-17

U s ing C a ll H u nt G r oups 4-19

Ov e rview 4-19

C o nfiguring a H u nt G r oup R u le 4-19

U s ing the C o ntact L i st Parameter 4-20

U s ing the A d ministration Web Server to C o nfigure H u nt G r oups 4-20

C l ient Station B l ind T r ansfers E x t ernal C a ller T o DID /H u nt G r oup 4-21

U s ing Shared L i ne A p pearance 4-22

M a naging C a ll Pickup 4-25

D i rected C a ll Pickup 4-25

G r oup C a ll Pickup 4-26

C a ll Park and Pickup 4-26

Multicast and G r oup Paging 4-27

Music O n H o ld 4-28

Ov e rview 4-28

C h anging the Internal Music Source 4-28

R e storing the D e fault Internal Music Source 4-29

U s ing a Streaming A u dio Server 4-30

U s ing the I V R w ith an S A S L i ne 4-30

Ex a m ple S A S w ith M O H 4-31

C o nfiguring the Streaming A u dio Server 4-32

E n hancements in R e lease 5.1 4-32

B r idge Mode 4-32

C a ll F o rward B r idge Mode 4-33

C a ll T r ansfer B r idge Mode 4-33

REGI S T ER E n hancement 4-33

R e new D HCP On S IP R e quest T i meout 4-34

C H A P T E R

C o nfiguring the L VS A u to-A t tendant 5-1

C o nfiguring A u to-A t tendant 5-1

H o w the A u to-A t tendant Works 5-1

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U s ing the I V R t o R e cord A u to-A t tendant Prompts 5-2

U s ing the Wizard to C o nfigure the A u to-A t tendant 5-4

U s ing the A d ministration Web Server to C o nfigure the A u to-A t tendant 5-6

D o wnloading Prompts 5-8

C o nfiguring D i al Plans for the A u to-A t tendant 5-9

A l ternative A A C o nfiguration 5-10

Switching B e tween A l ternative A As U s ing the I V R 5-10

X M L Scripting for the A u to-A t tendant 5-13

Ov e rview 5-13

X M L Scripting G r ammar 5-13

Node Type Di a log 5-14

M e nu T ype Di a log 5-14

D i alplan Statement 5-14

AA In s tructions 5-15

A u dio Instruction 5-15

A c tion Instruction 5-15

Noinput Instruction 5-15

Nom a tch I nstruction 5-16

M e nu Matched Instruction—R e cognition of T o uch T o ne (D MTP) K e y P resses 5-16

AA XML S cript E x a m ples 5-17

Ex a m ple 1—A A D e fault X M L Script 5-17

Ex a m ple 2 5-18

Ex a m ple 3—A A S cript with T w o T r eat m e nts 5-19

O f fice H o ur A A T r eat m e nt 5-19

Non-O f fice H o ur A A T r eat m e nt 5-20

A u to-A t tendant X M L In s tructions Set 5-22

C H A P T E R

SP A 9000 Fi e ld R e f e rence 6-1

Info T a b 6-2

Product Information 6-2

System Status 6-2

L i ne 1 /2 /3 /4 S tatus 6-3

FXS 1 /2 S tatus 6-3

A u to A t tendant Prompt Status 6-4

Internal Music Status 6-5

System T a b 6-6

System C o nfiguration 6-6

Miscellaneous Settings 6-6

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SIP T a b 6-7

SIP P a ra m e ters 6-7

SIP Tim e r V a lues (sec) 6-8

R e sponse Status C o de H a ndling 6-10

RTP P a rameters 6-10

S D P P a y load T yp e s 6-11

NAT Support Parameters 6-13

PBX P a ra m e ters 6-14

Internal Music Source Parameters 6-17

A u to A t tendant Parameters 6-18

PBX Phone P a ra m e ters 6-21

R e gional T a b 6-23

C a ll Progress T o nes 6-23

D i stinctive R i ng Patterns 6-24

D i stinctive C a ll Waiting T o ne Patterns 6-26

D i stinctive R i ng /C WT P a ttern N a m e s 6-26

R i ng and C a ll Waiting T o ne Spec 6-27

C o ntrol T i mer V a lues (sec) 6-28

V e rtical Service A c tivation C o des 6-29

V e rtical Service A n nouncement C o des 6-33

O u tbound C a ll C o dec Selection C o des 6-33

Miscellaneous 6-35

FXS 1 /2 T a b 6-38

L i ne E n able 6-38

N e twork Settings 6-38

SIP Settings 6-39

Subscriber Information 6-41

D i al Plan 6-41

M a ilbox Status 6-41

Streaming A u dio Server (S A S ) 6-42

C a ll F e ature Settings 6-43

A u dio C o nfiguration 6-43

FXS P o rt Polarity C o nfiguration 6-46

L i ne 1 /2 /3 /4 T a b 6-47

L i ne E n able 6-47

N e twork Settings 6-47

SIP Settings 6-48

Subscriber Information 6-50

D i al Plan 6-52

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Contents

NAT S e ttings 6-53

Proxy and R e gistration 6-54

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Preface

The LVS 9000 solution includes a line of IP communication products including desktop IP phones, an

IP PBX, and PSTN gateway

This guide describes basic administration and use of the Linksys SPA9000 IP PBX and the SPA400

PSTN gateway. It contains the following sections:

•

Document Audience, page xi

How This Document is Organized, page xii

Document Conventions, page xii

Related Documentation

The following documentation provides additional information about features and functionality of the

SPA9000:

•

LVS CTI Integration Guide

LVS Integration with ITSP Hosted Voicemail Guide

AA & IVR Quick Guides

SPA Provisioning Guide

SPA9000 User Guide

The following documentation describes how to use other Linksys Voice System products:

•

SPA900 Series IP Phones Administrator Guide

LVS Linksys Voice over IP Products Guide

SPA 2.0 Analog Telephone Adapter Administrator Guide

Technical Support

If you are an end user of LVS products and need technical support, contact the reseller or Internet

telephony service provider (ITSP) that supplied the equipment.

Technical support contact information for authorized Linksys Voice System partners is as follows:

•

LVS Phone Support (requires an authorized partner PIN)

888 333-0244 Hours: 4am-6pm PST, 7 days a week

E-mail support

[email protected]

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Technical Support

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C H A P T E R

Using the Linksys Voice System

This chapter provides an introduction to the components and functionality of the Linksys Voice System

(LVS). It includes the following sections:

•

The Linksys V o ice System, page 1-1

Technology Background, page 1-6

S P A 9000 Architecture, page 1-11

Where to Go From Here, page 1-16

The Linksys Voice System

This section provides basic information about the LVS VoIP PBX system and includes the following

topics:

•

Overview, page 1-1

S P A 400 SIP-PSTN Gateway and V o icemail, page 1-2

Auto-Attendant, page 1-3

S P A 9000 System Features, page 1-3

Additional Features When Used with SPA900 Series IP Phones, page 1-5

Overview

The Linksys Voice System (LVS) is an affordable, feature-rich, multi-line voice over IP (VoIP) telephone

system that provides sophisticated communication services to small business users. The LVS uses

standard TCP/IP protocols and can provide global connectivity through any Internet Telephony Service

Provider (ITSP) that supports Session Initiation Protocol (SIP). In addition, with the optional SPA400,

the LVS provides full interconnectivity with the Public Switched Telephone Network (PSTN).

The LVS solution, illustrated in Figure 1-1, provides a line of IP communication products that include

the following:

•

SPA9000 IP PBX

SPA400 SIP-PSTN gateway

SPA900 Series IP phones (SPA921, 922, 941, 942, and 962)

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Chapter 1 Using the Linksys Voice System

The Linksys Voice System

Figure 1-1 The Linksys Voice System (LVS) w ith the SPA9000 and SPA400

PSTN

Up to 4 FXO lines

Local voicemail

SPA400

SIP-PSTN

gateway

Switch

Internet

ITSP

ISP

SPA901, 921, 922, 941, 942, 962

SPA9000

IP PBX

FXS1

FXS2

Fax/Analog

Phones

The LVS 9000 system uses the power of VoIP to provide enterprise-quality telephony features to small

office/home office (SOHO) and small businesses. The LVS is based on open standards, such as SIP,. This

allows interoperation with other standards-based products and simplifies configuration and use. The

SPA9000, with a base license, supports up to four IP phones and up to 16 phones with an upgraded

license.

With the optional SPA400, the SPA9000 can also manage calls to and from the PSTN. The SPA9000 also

includes an Analog Telephone Adapter (ATA), with two FXS ports for connecting analog telephones, fax

devices, or an external music source for the music on-hold service included with the SPA9000.

The SPA9000 supports four independent line interfaces with numbers assigned by one to four different

ITSPs, with each line supporting up to 16 extensions. If the service provider supplies a group of

sequential direct inward dial (DID) phone numbers (such as 408-777-1000 through 777-1015) the

SPA9000 can support all the assigned numbers on a single line interface.

For information about LVS architecture, refer to the “S P A 9000 Architecture” section on page 1-11.

SPA400 SIP-PSTN Gateway and Voicemail

The SPA400 is optionally used with the SPA9000 to provide a SIP-PSTN gateway, providing voice

connectivity between the PSTN and local client stations connected to the SPA9000. It also provides a

local voicemail server.

Note

The SPA400 provides four FXO ports and occupies one line interface on the SPA9000.

A total of four SPA400 devices can be configured per SPA9000, using up to 16 analog phone lines and,

with the SPA9000, automatically routing calls to and from your existing PSTN telephone service.

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Chapter 1 Using the Linksys Voice System

The Linksys Voice System

Designed to be implemented with the SPA9000, the SPA400 lets cost-conscious business users take

advantage of all the high-value features on the LVS, which are typically found on much more expensive

voice communications systems. The SPA400 includes an integrated voicemail application supporting up

to 32 voicemail accounts with customized greetings, providing LVS users the ability to receive and

playback voicemail messages. The SPA400 ships with a USB voicemail module, which stores voicemail

prompts and allows recording of up to four hours of high-quality voice messages.

For detailed information about using the SPA400 voice services and voicemail servers, refer to

Chapter 3, “Configuring V o ice Service and V o icemail.”

Auto-Attendant

The Auto-Attendant is an internal service within the SPA9000. It plays pre-recorded voice messages that

offer the caller a menu of choices, so the Auto-Attendant can appropriately direct the call. After the

caller has made a choice, the call is routed to the appropriate extension. When the Auto-Attendant is

enabled, it parses and operates on user input (key presses or DTMF tones) following the rules specified

in the Auto-Attendant script on the SPA9000.

For detailed information about using and configuring the Auto-Attendant, refer to Chapter 5,

“Configuring the LVS Auto-Attendant.”

SPA9000 System Features

This following summarizes the features provided by the SPA9000:

•

SIP Application Server, Proxy, Registrar and Location Server (RFC3261)

Multiple Service Provider Lines / SIP Account Support (4)

Shared Line Appearance (SLA)

Configurable AA Answer Delay

Interactive Voice Response (IVR)

Recordable IVR Prompts

Automatic Call Distribution (ACD)

Configurable Call Routing

–

Least Cost Routing

Multiple DID Numbers Per VoIP Line

Call Routing to Multiple Extensions or Targeted User

Call Hunting - Sequential, Round Robin, Random

•

Phone Configuration and Management Server

–

Discovery and Configuration of IP Phones

Assignment of Extension

Assignment of Dial plan

Proxy Logging of SIP Messages

Phone Firmware Upgrade Management

•

Corporate Directory with Automatic Update

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Chapter 1 Using the Linksys Voice System

The Linksys Voice System

•

Configuration and Maintenance via Web Interface (Local or Remote

Status Display of All Connections

Remote Configuration via

–

•

–

HTTPS with XML Formatted Files

HTTP or TFTP with 256-Bit Encrypted Binary Files

•

Call Park -User Definable Parking Space Number

Call Unpark

Call Transfer - Attended and Blind

Call Forward

Group Paging

Intercom

Directed Call Pick Up

Group Call Pick Up

Music / Information via Streaming Audio Server (SAS) for Calls:

–

On Hold

Parked in the Parking Lot

Being Transferred

•

Simultaneous Ringing (Find Me Service)

Do Not Disturb

Voice Mail Integration - Service Provider Based

–

Voice Mail Notification via SUBSCRIBE / NOTIFY

Forward Call Directly to Voice mail

•

Integrated Media Proxy or Direct RTP Routing to ITSP

Differentiated Services (DiffServ) / Type of Service (TOS) Support

Two FXS (RJ-11) ports for Phones, Fax machines, Media Adapters

Voice encoding with G.711 (64kbit/s) and other codecs (G.723, G.726, and G.729

Fax Support using G.711 Pass-Through or T.38

Echo Cancellation (G.165)

Line Status - Active Line Indication, Name/Number

Digits Dialed with Number Auto-Completion

Call Hold

Call Waiting

Call Conferencing

Automatic Redial

Call Pick Up - Selective and Group

Call Forwarding - Unconditional, No Answer, On Busy

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Chapter 1 Using the Linksys Voice System

The Linksys Voice System

Additional Features When Used with SPA900 Series IP Phones

The following lists the additional features available when using the SPA9000 with SPA900 Series IP

phones:

•

Line Status - Active Line Indication, Name/Number

Digits Dialed with Number Auto-Completion

Call Hold

Call Waiting

Call Transfer - Attended and Blind

Call Conferencing

Automatic Redial

Call Pick Up - Selective and Group

Call Swap

Call Forwarding - Unconditional, No Answer, On Busy

Hot Line and Warm Line Automatic Calling

Call Log (60 entries each): Made, Answered, Missed Calls

Personal Directory with Auto-dial (100 entries)

Do Not Disturb

URI (IP) Dialing Support (Vanity Numbers)

On Hook Default Audio Configuration (Hands Free/Headset)

Multiple Ring Tones with Selectable Default Ring Tone per Line

Called Number with Directory Name Matching

Calling Number with Name - Directory Matching or via Caller ID

Subsequent Incoming Calls with Calling Name and Number

Date and Time with Intelligent Daylight Savings Support

Call Duration with Call Time Stamp Stored in Call Logs

Name/Identity (Text) Display at Start Up

Distinctive Ringing Based on Calling and Called Number

User Downloadable Ring Tones and Ring Tone Generator (Free from www.linksys.com)

Download on Demand Ring Tones - 10

Speed Dial Support

Configurable Dial/Numbering Plan Support - per Line

DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy

Syslog, Debug, Report Generation and Event Logging

Secure Call Encrypted Voice Communication Support

Built-in Web Server for Admin and Config with Multiple Security Levels

Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)

Require Admin Password to Reset Unit to factory Defaults Option

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Chapter 1 Using the Linksys Voice System

Technology Background

This section provides background information about the technology and protocols used by the SPA9000

system. It includes the following topics:

•

Session Initiation Protocol, page 1-6

S P A 9000 Media Proxy, page 1-7

Using the SPA9000 with a Firewall or Router, page 1-8

S P A 400 SIP-PSTN Gateway, page 1-8

Network Address Translation (N A T ), page 1-9

Session Initiation Protocol

The LVS is implemented using open standards, such as Session Initiation Protocol (SIP), allowing

interoperation with all ITSPs supporting SIP. Figure 1-2 illustrates a SIP request for connection to

another subscriber in the network. In the SIP protocol, the requestor of the session is called the user agent

server (UAS), while the receiver of the request is called the user agent client (UAC).

Figure 1-2 SIP Requests and Responses

SIP UA

SIP Proxy

RTP

SIP Proxy

SIP UA

Note

In this manual, the term client station is used to describe any SIP UA (including IP phones) that registers

with the SPA9000.

In a SIP VoIP network, when the SIP proxy receives a request from a client station (UAS) for a

connection and it does not know the location of the UAC, it forwards the message to another SIP proxy

in the network. Once the UAC is located and the response is routed back to the UAS, a direct peer-to-peer

session is established between the two UAs. The actual voice traffic is transmitted between UAs over

dynamically assigned ports using the Real-time Protocol (RTP).

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Chapter 1 Using the Linksys Voice System

Technology Background

In Figure 1-3, UserA and UserB are client stations (UAs) that register over the local area network to

which the SPA9000 PBX is connected. When UserA calls UserB, the SPA9000 acts as a SIP proxy and

establishes a session between the two UAs. After the session is established, RTP traffic flows directly

between the two client stations.

Figure 1-3 SPA9000 as a SIP Proxy

UserC

UserB

UserA

IP Router (firewall)

Hub/switch

Broadband modem

Internet

ISP

Internet (WAN)

Interface

SPA9000

ITSP

SIP Proxy with

media proxy enabled

When a user picks up the handset in an LVS system, the SPA9000 collects DTMF digits from a touchtone

analog telephone or the locally connected SPA900 Series IP phones. Unless the call is for a local client

station, the SPA9000 system sends the full number in a SIP INVITE message to another SIP proxy server

for further call processing.

To minimize dialing delay, a dial plan is maintained that is matched against the cumulative number

entered by the user. Invalid phone numbers that are not compatible with the dial plan are detected and

the user is alerted using a configurable tone (reorder) or announcement.

Figure 1-3 also illustrates connectivity between the SPA9000 and the ITSP over the Internet. When

UserA calls UserC, the SPA9000 directs the request to the SIP proxy at the ITSP, which is then

responsible for routing the request to UserC. Again, once the session is established, RTP packets are

exchanged directly between UserA and UserC. However, this requires that the firewall on the Internet

routers allow UserA access to the Internet. Because the SIP UAs are generally assigned IP addresses

dynamically through DHCP, this makes implementing a secure firewall policy more difficult.

SPA9000 Media Proxy

To address this possible security issue, the SPA9000 can also function as a media (RTP) proxy. This

option forces RTP traffic destined for the Internet (or IP WAN) to be directed to the SPA9000, which

then directs it to the remote UA. This configuration may simplify firewall configuration because the

client stations do not require direct access to the Internet through the firewall.

To enable the media proxy, set the PBX Parameters:<Force Media Proxy> parameter to True. With the

media proxy enabled, when UserA calls User C, the SPA9000 still acts as the SIP proxy and forwards

the request to the SIP server on the ITSP. However, even after the SIP session is established, the

SPA9000 continues to direct RTP packets between UserA and the ITSP.

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Technology Background

Local traffic is not affected by this configuration. When UserA initiates a call to UserB, RTP traffic still

flows directly between the two UAs. The media proxy only affects RTP traffic to a UA connected

through the ITSP.

Using the SPA9000 with a Firewall or Router

When using the SPA9000 behind a firewall or router, make sure that the following ports are not blocked:

SIP ports—By default, UDP port 5060 and 5061

RTP ports—16384 to 16482

Also disable SPI if this function exists on your firewall.

•

SPA400 SIP-PSTN Gateway

When a local user on the SPA9000 network initiates a call to a PSTN subscriber, the SPA400 acts as the

SIP-PSTN gateway, which converts the SIP and RTP media packets into the appropriate signal for

transmission to the PSTN switch. For example, if UserA calls UserD, the SIP request is routed by the

SIP proxy in the SPA9000 to the SPA400 (see Figure 1-4).

Figure 1-4 The SPA400 as a SIP-PSTN Gatew ay

PSTN

UserD

1 to 4 DID lines

SPA400

SIP-PSTN

Gateway

UserB

UserA

Switch

Internet (WAN) Interface

SPA9000

SIP Proxy

The SPA400 then converts the SIP and RTP packets it receives from UserA and the signals it receives

from the PSTN switch.

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Technology Background

Network Address Translation (NAT)

This section describes issues that arise when using the LVS on a network behind a network address

translation (NA) device. It includes the following topics:

•

N A T O verview, page 1-9

N A T Types, page 1-10

Simple Traversal of UDP Through N A T, p age 1-10

SIP-N A T I nteroperation, page 1-11

NAT Overview

Network Address Translation (NAT) allows multiple devices to share the same public, routable, IP

address for establishing connections over the Internet. NAT is typically performed by a router that

forwards packets between the Internet and the internal, private network.

A typical application of a NAT is to allow all the devices in a subscriber home network to access the

Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets

sent from the private network to the public network is substituted by NAT with the public IP address and

a port assigned by the router. The receiver of the packets on the public network sees the packets as

coming from the external address instead of the private address of the device.

The association between a private address and port and a public address and port is called a NAT

mapping. This mapping is maintained for a short period of time, that varies from a few seconds to several

minutes. The expiration time is extended whenever the mapping is used to send a packet from the source

device.

Figure 1-5 NAT Support w ith Session Border Controller Provided by ITSP

Private IP address

192.168.1.1

External IP address

assigned by ISP

192.168.1.101

192.168.1.102

NAT Device

Internet

ISP

DHCP

server

SPA9000

ITSP

SIP Proxy

192.168.1.100

Session Border

Controller

The ITSP may support NAT mapping using a Session Border Controller (see Figure 1-5). This is the

preferred option because it eliminates the need for managing NAT on the SPA9000. If this is not

available, you will need to discuss with the ITSP how to use the NAT Support Parameters provided by

the SPA9000, such as <Outbound Proxy> and <STUN Server Enable>.

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Technology Background

NAT Types

The different types of NAT implementation are sometimes divided into the following categories:

•

Full cone NAT—Also known as one-to-one NAT. All requests from the same internal IP address and

port are mapped to the same external IP address and port. An external host can send a packet to the

internal host, by sending a packet to the mapped external address

Restricted cone NAT—All requests from the same internal IP address and port are mapped to the

same external IP address and port. Unlike a full cone NAT, an external host can send a packet to the

internal host only if the internal host had previously sent a packet to it.

Port restricted cone NAT/symmetric NAT—Port restricted cone NAT or symmetric NAT is like a

restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can

send a packet to a particular port on the internal host only if the internal host had previously sent a

packet from that port to the external host.

With symmetric NAT all requests from the same internal IP address and port to a specific destination IP

address and port are mapped to a unique external source IP address and port. If the same internal host

sends a packet with the same source address and port to a different destination, a different mapping is

used. Only an external host that receives a packet can send a UDP packet back to the internal host.

Simple Traversal of UDP Through NAT

Simple Traversal of UDP through NATs (STUN) is a protocol defined by RFC 3489, which allows a

client behind a NAT device to find out its public address, the type of NAT it is behind, and the port

associated on the Internet connection with a particular local port. This information is used to set up UDP

communication between two hosts that are both behind NAT routers. Open source STUN software can

be obtained at the following website:

http://www.voip-info.org/wiki-Open+Source+VOIP+Software

STUN does not work with a symmetric NAT router. To determine the type of NAT your router uses,

complete the following steps:

Step 1

Enable debugging on the SPA9000:

1. Make sure you do not have firewall running on your PC that could block the syslog port (by default

this is 514).

2. On the administration web server, System tab, set <Debug Server> to the IP address and port number

of your syslog server.

Note that this address and port number has to be reachable from the SPA.

3. Set <Debug level> to 3 but you do not need to change the value of the <syslog server> parameter.

4. To capture SIP signaling messages, under the Line tab, set <SIP Debug Option> to Full. The output

is named syslog.514.log.

Step 2

Step 3

To determine the type of NAT your router is using set <STUN Test Enable> to yes.

View the syslog messages to determine whether your network uses symmetric NAT.

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SPA9000 Architecture

SIP-NAT Interoperation

In the case of SIP, the addresses where messages/data should be sent to a SPA9000 system are embedded

in the SIP messages sent by the device. If the SPA9000 system is sitting behind a NAT device, the private

IP address assigned to it is not usable for communications with the SIP entities outside the private

network.

Note

If the ITSP offers an outbound NAT-Aware proxy, this discovers the public IP address from the remote

endpoint and eliminates the need to modify the SIP message from the UAC.

The SPA9000 system must substitute the private IP address information with the proper external IP

address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer

address/port. For this, the SPA9000 system needs to perform the following tasks:

•

Discover the NAT mappings used to communicate with the peer.

This can be done with the help of an external device, such as a STUN server. A STUN server

responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP

address/port of the request, where the message contains the source IP address/port of the original

request. The SPA9000 system can send this request when it first attempts to communicate with a SIP

entity over the Internet. It then stores the mapping discovery results returned by the server.

Communicate the NAT mapping information to the external SIP entities.

If the entity is a SIP Registrar, the information should be carried in the Contact header that

overwrites the private address/port information. If the entity is another SIP UA when establishing a

call, the information should be carried in the Contact header as well as in the SDP embedded in SIP

message bodies. The VIA header in outbound SIP requests might also need to be substituted with

the public address if the UAS relies on it to route back responses.

Extend the discovered NAT mappings by sending keep-alive packets.

Because the mapping is alive only for a short period, the SPA9000 system continues to send periodic

keep-alive packets through the mapping to extend its validity as necessary.

SPA9000 Architecture

This section describes the basic architecture, function, and configuration options for the SPA9000. It

includes the following topics:

•

Architectural Components, page 1-12

Multicast Addressing and Group Paging, page 1-13

Configuration Options, page 1-14

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SPA9000 Architecture

Architectural Components

Figure 1-6 SPA9000 Architecture

SPA9000

Application

server

Administration

web server

(408)111-1000 to 7

ITSP SIP Proxy

Line 1

Line 2

(408)111-1111

SPA 400

SIP-PSTN

gateway

(408)111-1112

(408)111-1113

(408)111-1114

PSTN

SIP Proxy

SIP Registrar

(949)111-2000 to 7

(888)111-3000 to 7

ITSP SIP Proxy

Line 3

Line 4

Media (RTP) Proxy

Call

FXS1

FXS2

imusic

park

As shown in Figure 1-6, the SPA 9000 provides four logical line interfaces, referred to as Line 1, 2, 3,

and 4. Each line can be configured with the same or a different ITSP. Each SPA400 also occupies one

line interface. The SPA9000 has five internal clients that register implicitly with the internal SIP proxy:

•

FXS1 (fxs1)

FXS2 (fxs2)

Call Park (callpark)

Auto-Attendant (aa)

Internal Music Server (imusic)

FXS1 and FXS2 correspond to the two physical FXS ports. The FXS ports can only register with the

local SIP proxy. The Call Park is used to maintain calls that are parked, and AA is a scriptable

auto-attendant application.

Table 1-1 Architectural Com ponents

Architectural Component

Function

SIP proxy and Registrar server

Accepts registration from client stations and

proxies SIP messages.

Media proxy server

Configuration server

Application server

Proxies RTP packets between client stations and

proxies SIP messages.

Serves configuration files to client stations and

auto configures un-provisioned client stations.

Supports advanced features such as call

park/pickup, directory, directed call pickup and

group paging, hunt groups, and shared line

appearances.

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SPA9000 Architecture

Table 1-1 Architectural Com ponents

Architectural Component

Function

Internal music source

Streams audio files to client stations (both on-net

and off-net).

The FXS1 and FXS2 can optionally be connected

to an external music source to act as a streaming

audio server (SAS). When working in this mode,

each FXS port can handle up to 10 concurrent

calls.

Administration web server

ATA with 2 FXS ports

Allows configuration and monitoring of the

SPA9000.

Each FXS port can be connected to analog

phones, fax machine, or an external music

source. Each port can support up to two calls

simultaneously. The FXS ports can only register

to the internal proxy server.

Call park

The call park is used to maintain calls that are

parked and can handle up to 10 calls

simultaneously

Auto-Attendant

AA is a scriptable auto-attendant application that

can handle up to 10 calls simultaneously

Multicast Addressing and Group Paging

The <Multicast Address> parameter on the PBX Parameters page defines the multicast address used by

the SPA9000 and the SPA900 Series phones to communicate with each other. The default value is

224.168.168.168:6061. This address can also be set using the IVR option 181 and reviewed using option

180.

The <Group Page Address> parameter is used by the SPA9000 for group paging of all active client

stations. The default value is 224.168.168.168:34567.

The SPA9000 can send the following messages to the phone group:

•

Graceful Reboot

Immediate Reboot

Graceful Restart

Immediate Restart

Group Page Start

Group Page End

Get Ringing Calls

Client stations send multicast messages to the SPA9000 when they are looking for the configuration

server.

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SPA9000 Architecture

Configuration Options

This section describes the different methods for configuring the SPA9000. It includes the following

topics:

•

Interactive V o ice Response, page 1-14

Setup Wizard

Administration Web Server

Local Client Configuration and Registration, page 1-15

Remote Provisioning of the SPA9000, page 1-15

Interactive Voice Response

The Interactive Voice Response (IVR), which is strictly for administration purposes, lets you use an

analog phone to perform basic configuration and troubleshooting operations. To access the IVR, connect

an analog phone to an FXS port and press **** to access the IVR menu.

For detailed information about using the IVR, refer to the “Using the Interactive V o ice Response

Interface” section on page 2-8. A convenient quick-reference for the IVR is available at the following

website:

http://www.linksys.com/

Setup Wizard

The Setup Wizard is a convenient way to perform initial configuration for the SPA9000. It provides

step-by-step guidance for configuring the basic operation of voice services, voicemail, and most of the

main features provided by the SPA9000.

The Setup Wizard overwrites any existing configuration information that has been entered through the

administration web server, so it should not be used for ongoing administration and configuration unless

it provides adequate functionality for you to use it exclusively. You can, however, use the Setup Wizard

for initial configuration and then use the administration web server for ongoing configuration and

maintenance.

You can download the latest Setup Wizard from the following website:

http://www.linksys.com/

To start the Wizard, just double-click on the executable file. For information about getting started with

the Setup Wizard, refer to the “Using the Wizard for Initial Configuration” section on page 2-13

Administration Web Server

The administration web server provides a series of web pages that let you enter detailed configuration

information for the many features and options provided by the SPA9000. It also lets you monitor the

status of the attached client stations.

The administration server provides a basic and an advanced view from which the various configuration

parameters can be accessed. The Provisioning tab is only visible from the advanced Administrator

account view of the web interface.

To access the administration web server, direct a browser to the Internet (WAN) interface of the

SPA9000. To determine this address, use IVR Option 110#.

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SPA9000 Architecture

For detailed information about using the administration web server, refer to the “Using the

Administration Web Server” section on page 2-22. For a description of each parameter provided by the

administration web server, refer to Chapter 6, “SPA9000 Field Reference”

Local Client Configuration and Registration

SPA9000 provides a TFTP server to assign configuration information to the locally attached client

stations. When the SPA9000 receives a request for /cfg/init_$MA.xml, it automatically assigns the next

available user id (extension number) to the client station. The initial user ID is configured using the PBX

Phone Parameters:<Next Auto User ID> parameter and is automatically incremented each time a new

number is assigned. Before assigning a new user ID, the SPA9000 also checks if there is any registered

client station using that ID and increases the ID until an unused value is found.

Remote Provisioning of the SPA9000

The SPA9000 provides for secure provisioning and remote upgrade. Provisioning is achieved through

configuration profiles transferred to the device via TFTP, HTTP, or HTTPS. User intervention is not

required to initiate or complete a profile update or firmware upgrade.

The SPA9000 can be configured to automatically resync its internal configuration state to a remote

profile periodically and during power up. The automatic resync is controlled by configuring the profile

URL for the device.

The SPA9000 accepts profiles in XML format, or alternatively in a proprietary binary format, which can

be generated by a profile compiler tool available to qualified VoIP vendors and partners from Linksys.

The SPA9000 supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of the

profile encryption key (initial provisioning stage), the SPA9000 can receive a profile from an encrypted

channel (HTTPS with client authentication), or it can resync to a binary profile generated by the

Linksys-supplied profile compiler. In the latter case, the profile compiler can encrypt the profile

specifically for the target SPA9000, without requiring an explicit key exchange.

Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades using HTTPS are not

supported). Remote upgrades are controlled by configuring the desired firmware image URL into the

SPA9000 via a remote profile resync.

For further information about remote provisioning refer to the LVS SPA Provisioning Guide.

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Where to Go From Here

The following table summarizes the steps required to implement and configure the SPA9000 system and

indicates where to look for the information required.

Task

Refer to

Establishing connectivity among system

components

Chapter 2, “Getting Started.”

Configuring voice services and SPA400 or

ITSP-hosted voicemail

Chapter 3, “Configuring V o ice Service and

V o icemail.”

Configuring and understanding SPA9000 features Chapter 4, “Configuring SPA9000 Features.”

Configuring or writing XML scripts for the

Auto-Attendant

Chapter 5, “Configuring the LVS

Auto-Attendant.”

Identify the function or setting required for a

specific parameter on the administration web

server pages

Chapter 6, “SPA9000 Field Reference”

For additional information about specific functionality or features of the SPA9000, refer to the following

documents:

•

LVS CTI Integration Guide

LVS Integration with ITSP Hosted Voicemail Guide

Auto-Attendant Quick Guide

Interactive Voice Response (IVR) Quick Guide

SPA Provisioning Guide

SPA900 Series IP Phones Administrator Guide

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C H A P T E R

Getting Started

This chapter provides the information required to implement a SPA9000 system, including making the

required network connections and establishing basic connectivity among the system components. It

includes the following sections:

•

Implementing L V S, page 2-1

Using the Interactive V o ice Response Interface, page 2-8

Initial Setup and Configuration, page 2-12

Setting the S P A 9000 Administrator Account Password, page 2-22

Using the Administration Web Server, page 2-22

Advanced Methods of Configuration, page 2-24

Client Registration, page 2-30

Troubleshooting and Configuration FAQ, page 2-31

Implementing LVS

This section describes the first steps in implementing a SPA9000 system (LVS). It includes the following

topics:

•

Using the SPA9000 and SPA400 as a V o IP PBX System, page 2-2

Using the LVS as a Key System, page 2-3

S P A 9000 Hardware, page 2-4

S P A 400 Hardware, page 2-6

Bandwidth Requirements, page 2-6

Caring for Y o ur Hardware, page 2-7

Making the Physical Connections, page 2-8

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Using the SPA9000 and SPA400 as a VoIP PBX System

Figure 2-1 illustrates the hardware required to implement a SPA9000 system with the SPA400 connected

to the PSTN.

Figure 2-1 SPA9000 and SPA400 System Hardw are

PSTN

PC connected

through SPA9x2 IP phone

Up to 4 FXO lines

SPA400

(optional)

IP Router/

Broadband

Modem

Hub/switch

Public

Internet

ISP

SPA901, 921, 922, 941, 942, 962

Internet (WAN) Interface

SPA9000

ITSP

The following are the basic hardware requirements to implement an LVS PBX system:

•

SPA9000

One or more SPA900 series IP Phones (as client stations)

Ethernet network cables

One or more Ethernet switches with QoS support and with the required number of available ports

A PC for configuration of the SPA9000 and other client stations on the network. This can be

connected to the switch or directly to the Ethernet port on the SPA922, 942, or 962 IP phone.

The following components can optionally be added to provide additional functionality:

•

FAX machine to send or receive faxes

A router with QoS support, and a broadband (cable/DSL) modem (gateway) with connectivity to an

ISP

•

SPA400 SIP-PSTN gateway for connectivity to the PSTN and local voicemail service

PSTN DID lines

For best results, use a router and switch that both support Quality of Service (QoS). QoS allows top

priority to be assigned to voice traffic. Otherwise, the quality of the voice connection may suffer when

large files are moved over the network.

Note

The SPA9000, the SPA400, and the SPA900 Series IP phones should all be within the same LAN.

Separating the devices through VPN encryption, firewalls, routers, or other devices that affect multicast

traffic may prevent proper functioning of the IP PBX system.

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In addition, you need at least one active ITSP phone service account and its settings (including DID

number for incoming calls) if you use the Internet for telephone service. If you are using the SPA400 for

connection to the PSTN, you need at least one active PSTN line.

The LVS can also be implemented with hosted voicemail services provided by the ITSP. In this scenario,

the SPA-400 is not configured for voicemail, but is only used for as a SIP-PSTN gateway, providing

telephone connectivity to PSTN subscribers.

Note

For first-time installation of the SPA9000, it is recommended that you use the SPA9000 Setup Wizard,

which you can download from www.linksys.com. For further information, see “Using the Wizard for

Initial Configuration” section on page 2-13.

Using the SPA9000 as a Media Proxy

The SPA9000 can act as a media proxy, which means that all RTP traffic between local client stations

and client stations on the Internet are routed through the SPA9000. This simplifies firewall configuration

because only the SPA9000 requires access to the Internet through the firewall.

By default, the SPA9000 acts only as a SIP proxy. This means that once a SIP connection is established,

further communication between the SIP UAs occurs directly. To enable the SPA9000 as a media proxy,

set the PBX Parameters:<Force Media Proxy> parameter to yes.

Using the LVS as a Key System

An easy and cost-effective way to implement the SPA9000 is as a replacement of a legacy key system,

in which a single DID line from the PSTN supports a single extension. In a key system all lines appear

as shared lines. This allows a user to pick up a ringing line from any phone in the key system (see

Figure 2-2).

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Figure 2-2 Using LVS w ith SPA9000 and SPA400 as a Key System

PSTN

Up to 4 DID lines per SPA400

1 to 4 SPA400s

Shared line appearance

Hub/switch

Internet

ISP

IP Router/

Broadband modem

Internet (WAN) Interface

SPA9000

ITSP

When implementing the SPA9000 and SPA400 as a key system, the SPA9000 is connected through a

switch to one or more SPA400s, which are then connected to the existing DID lines from the PSTN. The

SPA9000 has four line interfaces, each of which can support one SPA400 or a single VoIP account with

an ITSP. A single VoIP account with an ITSP can map to multiple DID numbers assigned by the ITSP.

Each SPA400 can support up to four DID lines assigned by the PSTN, so with a SPA400 connected to

all four line interfaces, the SPA9000 can support up to 16 DID lines to the PSTN.

SPA9000 Hardware

The following are the ports provided by the SPA9000:

•

Two analog (FXS) telephone ports (Phone 1 and Phone 2), which are designed for use by the

following devices:

–

Analog telephone

Fax machine

Music/audio player with a music source adapter (RJ11-to-Line-In) for use as a Streaming Audio

Server (SAS)

•

Two Ethernet ports, designed for the following functions:

–

Ethernet: administrative access for troubleshooting with a directly connected PC or laptop

Internet: SIP call traffic and signaling to client stations and administration web server access

(through switch)

SPA9000 Back Panel

The SPA9000 ports are located on the back panel.

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Figure 2-3 SPA9000 Back Panel

The following are the interfaces provided by the SPA9000, from left to right:

•

Phone 1/2—Connect to an analog telephone or fax machine with an RJ-11 cable.

Internet—Connect to a switch, router, or broadband (cable/DSL) modem. Also referred to as the

WAN port, because it provides connectivity to the wide area VoIP network.

•

Ethernet—For troubleshooting only.

Power—Connect to the power adapter.

The Front Panel

The SPA9000 LEDs are located on its front panel.

Figure 2-4 SPA9000 Front Panel

The following are the LEDs provided by the SPA900, from left to right:

•

Power—Steady green: powered on and connected to the Internet. Flashing: not connected to the

Internet or is booting or upgrading firmware.

•

Ethernet—Steady green: active connection. Flashing: indicates traffic.

Phone 1/2—Steady green: active/registered connection to ITSP through port. Flashing: in use or off

hook.

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SPA400 Hardware

Figure 2-5 SPA400 Back Panel

The SPA400 provides the following interfaces (from left to right):

•

USB—Use this to connect the SPA400 USB voicemail module containing voicemail prompts and

provides the storage location for saving voice mailbox messages.

•

Ethernet—Connect to the SPA9000 through the appropriate switch.

Line 1 to 4—Connect to the telephone line provisioned by your PSTN provider.

Power—Connect to the power supply.

Figure 2-6 SPA400 Front Panel

The SPA400 provides the following LEDs, from left to right:

•

Power—Steady green: powered on and connected to the Internet. Flashing: not connected to the

Internet, booting, or upgrading firmware.

•

Status—Steady green: SPA400 registered to the SPA9000. Flashing: SPA400 not registered.

Line 1-4—Steady green: line is active; flashing: ringing; off: idle.

Ethernet—Steady green: active connection. Flashing: indicates traffic.

USB—Steady green: USB voicemail module registered. Off: no module detected.

Bandwidth Requirements

Depending on how you have your IP phones configured, each call requires 55 to 110 kbps in each

direction. Therefore, using G.729 as the voice codec setting, and with an average business-grade

broadband Internet connection supporting 1.5 Mbps downstream and 384 kbps upstream, a total of seven

(7) simultaneous conversations can be reliably supported with adequate bandwidth available for file

downloads.

Linksys recommends using the SPA9000 with QoS-capable networking equipment that can prioritize the

VoIP application traffic. QoS features are available on many Linksys data networking switches and

routers. A QoS-enabled router prioritizes the packets going upstream to the ISP. Table 2-1 illustrates the

bandwidth budget using different codecs.

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Table 2-1 Ethernet Bandw idth Budget for Off-Net VoIP Calling

Approximate bandwidth budget for each

side of conversation

Codec

2 calls

4 calls

6 calls

8 calls

G.711

110 kbps

220 kbps 440 kbps 660 kbps 880 kbps

174 kbps 348 kbps 522 kbps 696 kbps

158 kbps 316 kbps 474 kbps 632 kbps

142 kbps 284 kbps 426 kbps 568 kbps

126 kbps 252 kbps 378 kbps 504 kbps

110 kbps 220 kbps 330 kbps 440 kbps

G.726-40 87 kbps

G.726-32 79 kbps

G.726-24 71 kbps

G.726-16 63 kbps

G.729

55 kbps

This table is based on the following assumptions:

•

Bandwidth Calculated with No Silence Suppression

20 Millisecond of payload per RTP packet

Note

The use of silence suppression can reduce the average bandwidth budget by 30% or more.

For more information about bandwidth calculation, refer to the following websites:

http://www.erlang.com/calculator/lipb/

http://www.packetizer.com/voip/diagnostics/bandcalc.html

Caring for Your Hardware

The SPA9000, SPA400, and the SPA900 Series phones are electronic device that should not be exposed

to excessive heat, sun, cold or water. To clean the equipment, use a slightly moistened paper or cloth

towel. Do not spray or pour cleaning solution directly onto the hardware unit.

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Using the Interactive Voice Response Interface

Making the Physical Connections

To establish or verify the necessary connectivity complete the following steps.

Note

This includes the steps required to establish basic connectivity for LVS and does not address the

configuration of other networking components that may affect VoIP services.

To complete these steps, you need a multiport switch, Ethernet cables, the SPA9000, and a PC connected

to the network.

1. Connect a network cable between the SPA9000 Internet Port and the switch.

2. Connect a network cable between the computer you use to configure the SPA9000 and the switch.

3. Plug an analog phone into the Phone 1 port on the SPA9000.

4. Connect the included Power Adapter to the power port of the SPA9000.

The status LED starts flashing as the SPA9000 boots up.

5. Connect a network cable from the broadband modem to the Internet port of the broadband router.

6. Connect an network cable from one of the LAN ports of the router to the Uplink port of the LAN

switch.

7. Connect the power adapters to the power ports of the broadband router and the LAN switch.

8. If the SPA9000 has been used previously, reset it to its factory defaults before starting configuration.

a. Connect an analog phone to Phone 1 or Phone 2.

b. Press **** to access the IVR menu.

For detailed information about the IVR system, refer to “Using the Interactive Voice Response

Interface” section on page 2-8.

c. Press 73738# and then 1#.

This resets the unit to its factory defaults.

d. Wait about 30 seconds while the system reboots.

Using the Interactive Voice Response Interface

This section describes how to read or write basic network configuration settings using a touchtone

telephone connected to one of the FXS (RJ-11) phone ports of the SPA9000. It includes the following

topics:

•

Using the IVR Menu, page 2-9

IVR Options, page 2-10

Entering a Password through the IVR, page 2-12

By default, there is no password required for any of the IVR options. If the Administrator account

password is set, password authentication is required for some options.

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Using the Interactive Voice Response Interface

The interactive voice response (IVR) features that are available depend on your system configuration.

A convenient quick-reference for the IVR is available at the following website:

http://www.linksys.com/

Using the IVR Menu

To use the IVR menu, complete the following steps.

Step 1

Connect an analog telephone to the Phone 1 or Phone 2 port of the SPA9000.

Note

You can only access the IVR menu through an analog telephone, not an IP phone.

Step 2

Press **** (quickly press the star key four times).

Wait until you hear “Linksys configuration menu.”

Note

You cannot access the IVR from a phone that is currently connected to a call.

Step 3

Step 4

Refer to Table 2-2 to identify the option required.

Enter the required option followed by the # (pound) key.

To enter a period, use the star key (*).

When entering a value, such as an IP address, to exit without entering any changes, press the * (star) key

twice within half a second. Otherwise, the * is treated as a decimal point.

After entering a value, such as an IP address, press the # (pound) key to indicate you have finished your

selection. To save the new setting, press 1. To review the new setting, press 2. To re-enter the new setting,

press 3. To cancel your entry and return to the main menu, press * (star).

For example, to enter the IP address 191.168.1.105 by keypad, press these keys: 191*168*1*105. Press

the # (pound) key to indicate that you have finished entering the IP address. Then press 1 to save the IP

address or press the * (star) key to cancel your entry and return to the main menu.

If the menu is inactive for more than one minute, the SPA9000 times out. You need to re-enter the menu

by pressing ****.

Step 5

To exit the menu, hang up the telephone.

The settings you have saved take effect after you hang up the telephone. The SPA9000 may reboot at this

time.

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Using the Interactive Voice Response Interface

IVR Options

Table 2-2 summarizes the options provided by the IVR.

Table 2-2 IVR Options

IVR Action

IVR Menu Choice

Parameter(s)

Notes

Enter IVR Menu

* * * *

None

Ignore SIT or other tones until you hear,

“Linksys configuration menu. Please

enter option followed by the pound key or

hang-up to exit.”

Exit IVR Menu

Check DHCP

3948

100

None

IVR announces if DHCP in enabled or

disabled.

Enable/Disable DHCP

101

Enter 1 to enable

Enter 0 to disable

Requires password

Check WAN IP Address 110

None

IVR announces the current IP address of

the WAN port.

Set Static IP Address

111

Enter IP address using

numbers on the telephone

DHCP must be “Disabled,” otherwise you

hear, “Invalid Option,” if you try to set

key pad. Use the * (star) key this value.

when entering a decimal

Requires password

point.

Check Network Mask

Set Network Mask

120

121

None

IVR announces the current network mask

of SPA.

Enter value using numbers on DHCP must be “Disabled,” otherwise you

the telephone key pad. Use hear, “Invalid Option,” if you try to set

the * (star) key when entering this value.

a decimal point.

Requires password

Check Static Gateway IP 130

Address

None

IVR announces the current gateway IP

address of SPA.

Set Static Gateway IP

Address

131

Enter IP address using

numbers on the telephone

DHCP must be “Disable,” otherwise you

hear, “Invalid Option,” if you try to set

key pad. Use the * (star) key this value.

when entering a decimal

Requires password

point.

Check MAC Address

140

None

IVR announces the MAC address of SPA

in hex string format.

Check Firmware Version 150

None

IVR announces the version of the

firmware running on the SPA.

Check Primary DNS

Server Setting

160

IVR announces the current setting in the

<Primary DNS> parameter.

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Table 2-2 IVR Options (continued)

Set Primary DNS Server 161

Enter IP address using

numbers on the telephone

key pad. Use the * (star) key

when entering a decimal

point.

Requires password

Check administration web 170

server port

None

IVR announces the port that the web

server is listening on. (Default is 80)

Check LAN IP Address

210

180

181

None

IVR announces the current IP address of

the LAN port.

Check PBX multicast

address

None

IVR announces the current value.

Set PBX multicast

address

Enter IP address and port.

Use * key for entering a dot. Port fields. Requires Password

For example,

Enter a * between the IP address and the

224.168.168.169:8089 is

224*168*168*169*8089.

Enable/Disable

administration web server

7932

Enter 1 to enable

Enter 0 to disable

Requires password

Manual Reboot of Unit

732668

877778

None

After you hear “Option Successful,” hang

up. Unit reboots automatically.

User Factory Reset of

Unit

Enter 1 to confirm

Enter *(star) to cancel

operation

SPA prompts for confirmation. After

confirming, you hear “Option

Successful.” Hang up. Unit reboots and

all “User Changeable” configuration

parameters are reset to factory default

values.

WARNING:

ALL “User-Changeable”

NON-DEFAULT

SETTINGS WILL BE

LOST!

This might include

network and service

provider data.

Factory Reset of Unit

WARNING:

73738

Enter 1 to confirm

Enter * (star) to cancel

operation

SPA prompts for confirmation. After

confirming, you hear “Option

Successful.” Hang up. Unit reboots and

all configuration parameters are reset to

factory default values.

ALL NON-DEFAULT

SETTINGS WILL BE

LOST!

This includes network and

service provider data.

Note

The items marked with “Requires Password” only require a password if the Administrator password is

set.

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Initial Setup and Configuration

Entering a Password through the IVR

To input the password using the phone keypad, the following translation conventions apply:

–

To input: A, B, C, a, b, c—press “2’

To input: D, E, F, d, e, f—press “3’

To input: G, H, I, g, h, i—press “4’

To input: J, K, L, j, k, l— press “5’

To input: M, N, O, m, n, o—press “6’

To input: P, Q, R, S, p, q, r, s—press “7’

To input: T, U, V, t, u, v—press “8’

To input: W, X, Y, Z, w, x, y, z—press “9’

To input all other characters in the Administrator account password, press “0’

Note

This translation convention only applies to the password input.

For example, to input password test#@1234 by phone keypad, you need to press the following

sequence of digits: 8378001234.

1. After entering a value, press the # (pound) key to indicate end of input.

–

To save value, press 1.

To review the value, press 2.

To re-enter the value, press 3.

To cancel the value entry and return to the main configuration menu, press *’ (star).

Notes:

–

The final # key is not included in the password value.

Saved settings take effect when the telephone is hung-up, and if necessary, the SPA9000

automatically reboots.

2. After one minute of inactivity, the unit times out. The user needs to re-enter the configuration menu

from the beginning by pressing * * * *.

Initial Setup and Configuration

This section describes how to complete the initial connection and configuration of the SPA9000 system.

It includes the following topics:

•

Licensing, page 2-13

Using DHCP or Static IP Addressing, page 2-13

Using the Wizard for Initial Configuration, page 2-13

Using the Wizard to Upgrade Software, page 2-18

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Initial Setup and Configuration

Licensing

The SPA9000 is shipped from the factory with a four-user license. This means that the SPA9000 allows

registration from up to four external IP addresses (first come/first serve). If you need to support more

client stations, you can purchase a 16-user license and install it by entering the license key using the

<License Keys> parameter on the Provisioning page. To obtain a license, contact

[email protected].

Using DHCP or Static IP Addressing

Before running the Setup Wizard, you need to decide whether you are using DHCP or static IP

addressing. Static IP addressing is recommended for the SPA400. The SPA900 series phones typically

obtain their addresses through the DHCP server on the broadband router that connects to the broadband

modem. The SPA9000 can either be assigned a static address if this has been provided by your ISP, or it

can be assigned a dynamic IP address by the DHCP server on your router or by the ISP.

If you are using static IP addressing for the SPA400, you need to know the correct static IP address to

assign. To determine the dynamically assigned address of the SPA400, you can use the Setup Wizard or

view the DHCP client table for the broadband router.

To determine the current IP address of the SPA9000, complete the following steps.

Step 1

Step 2

Step 3

Connect an analog telephone to one of the FXS (RJ-11) ports on the SPA9000.

Press **** on the keypad to access the IVR menu.

Press 110# and note the response.

Using the Wizard for Initial Configuration

Step 1

After completing the required physical network connections and powering on all devices, start the SPA

Setup Wizard.

The latest wizard can be obtained at the following URL:

www.linksys.com/

Figure 2-7 shows the first screen that you should see.

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Figure 2-7 SPA Setup Wizard—Page 1

Step 2

Click Next.

The subsequent screen provides a review of the setup requirements that are required to successfully

complete the wizard. After verifying that you have completed the required setup, click Next.

The system displays Page 3 of the wizard (see Figure 2-8).

Figure 2-8 SPA Setup Wizard—Page 3

Step 3

Note

If you are configuring the SPA9000 for the first time, accept the default and click Next on Page 3.

If you are using the wizard to change your existing configuration, click the second radio button on Page

3 and enter the IP address of your SPA9000 on the page that appears.

If you use the Setup Wizard to configure the SPA9000 after using the administration server web, you

may lose any changes to the factory default configuration that you made using the administration server

web pages.

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Initial Setup and Configuration

Complete the following steps before continuing with the wizard:

1. Connect a multi-port switch to the router in your network.

2. Connect the Internet port on the SPA9000 to the switch.

3. Connect the SPA900 series phones to the switch.

4. Connect the administration PC to the switch.

5. Ensure that all devices are powered up.

6. Connect the SPA400s to the switch.

7. Power on the SPA400s.

8. Connect the FXO ports of the SPA400 to the phone ports connected to the PSTN (demarc) using a

standard telephone cord.

9. Click Next on Page 4 of the wizard.

Step 4

After making the necessary connections, click Next on Page 5 of the wizard.

The displays Page 6, as shown in Figure 2-9.

Figure 2-9 SPA Setup Wizard—Page 6

Step 5

Complete Page 6:

1. Select the MAC address of the SPA400 from the pull-down selection list.

2. Type the Administrator account password if it is set.

Note

The administrator user ID for the SPA400 is Admin (with a capital A). By default, no password

is required (leave the password field empty). The administrator user ID for the SPA9000 is

admin (with a lower-case a).

3. Select the connection type (DHCP or Static IP) from the pull-down selection list.

If using DHCP, skip Step d. and do not enter any IP addresses.

Note

It is highly recommended that you use Static IP addressing for the SPA400.

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4. To use Static IP, type the static IP address, type the default router address (in the Gateway field),

and type the addresses of the primary and secondary DNS servers.

The Secondary DNS server is optional when using static IP addressing.

5. Click Submit on Page 6.

Step 6

Complete the following steps, as described on Page 7 and Page 8 of the wizard:

1. Check the firmware version.

a. Plug an analog phone into the Phone 1 port of the SPA9000.

b. Pick up the analog phone and enter **** on the phone keypad to access the IVR menu.

c. Press 150# to hear the firmware version.

2. Enable web access.

a. Enter 7932# on the phone keypad.

This allows you to enable or disable the administration web server on the SPA9000.

b. Press 1#.

This enables the administration web server.

3. Press Next on Page 7.

4. If it has been used before, reset the SPA9000 to the factory defaults:

a. On the keypad of the analog phone, press 73738#.

This allows you to reset the unit to its factory default.

b. Press 1 on the keypad.

5. If it has been used before, reset each SPA900 series IP phone to its factory defaults.

a. Press the Menu button on the SPA900 series phone.

b. Use the Navigate button to scroll down to Factory Reset.

(or press Menu, type 14, and click OK).

c. Click Select.

Note

To reset a SPA901 phone press **** on the SPA901 keypad, press 73738#, and then press 1.

6. Press Next on Page 8.

The system displays Page 9, shown in Figure 2-9.

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Figure 2-10 SPA Setup Wizard—Page 9

Step 7

Type the IP address of the SPA9000 on the Internet.

To determine the Internet IP address of the SPA9000:

1. Pickup the handset of an analog phone connected to the Phone 1 port of the SPA9000.

2. Press ****.

3. Press 110# and note the address.

Step 8

Step 9

Type the Administrator account password on Page 9 if this has been supplied to you. Otherwise, leave

this field blank.

Click Next.

The system displays Page 10 of the Setup Wizard, which is the last page required to establish basic

connectivity for the system (see Figure 2-11).

Figure 2-11 SPA Setup Wizard—Page 10

Step 10 Select the connection type (DHCP or Static IP) from the pull-down selection list.

Step 11 If you are using Static IP, type the IP addresses of the primary and secondary DNS servers.

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Initial Setup and Configuration

Step 12 Click Submit.

The system displays the Main Menu of the Setup Wizard (see Figure 2-12).

Figure 2-12 SPA Setup Wizard—Main Menu

You can use the options on this menu to perform the following configuration operations:

•

Software Upgrade—Upgrade the SPA9000, SPA400, or SPA900 series telephone software (see the

“Using the Wizard to Upgrade Software” section on page 2-18).

Configure SPA400—Configure PSTN voice services and voicemail services using the SPA400 (see

the “Using the S P A 9000 Setup Wizard to Configure the SPA400” section on page 3-8.)

Configure SPA9000—Configure voice services and voicemail with an ITSP (see the “Using the

Wizard to Configure ITSP V o ice Services and V o icemail” section on page 3-1).

Configure Client Stations—Configure the SPA900 series clients (see the “Configuring Client

Stations” section on page 4-2).

Configure Advanced Features:

–

Change the Administrator password (see the “Setting the SPA9000 Administrator Account

Password” section on page 2-22).

Hunt groups, shared extensions, and other features (see the “Using the Wizard to Configure

S P A 9000 V o ice Features” section on page 4-1)

Auto-Attendant (see the “Using the Wizard to Configure the Auto-Attendant” section on

page 5-4).

Using the Wizard to Upgrade Software

Step 1

Download and save the latest software for whatever devices you want to upgrade from the following

website:

www.linksys.com/upgrade

Step 2

Step 3

Make a note of the folder where you save it locally because you need this information during the upgrade

process.

Select Software Upgrade on the Main Menu (Figure 2-12).

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The system displays the first page of the Upgrade Wizard (see Figure 2-13).

Figure 2-13 Softw are Upgrade Wizard—Page 1

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Step 4

Select one of the following types of devices to upgrade:

•

SPA941/921

SPA942/922

SPA962

SPA901

SPA9000

SPA400

Step 5

Step 6

Click Next.

The system displays Page 2 of the Software Upgrade Wizard (see Figure 2-14

Figure 2-14 Softw are Upgrade Wizard—Page 2

Step 7

Type the IP address of the device you are upgrading and enter the Administrator account password, if

this is set.

1. To determine the IP address of a SPA92x or 94x phone, press the Menu button,

2. Scroll down to Network and press the left button (Select).

3. Note the Current IP address, and enter it into Page 2 of the Software Upgrade wizard.

To determine the IP address of a SPA901 phone, pickup the handset, press ****, followed by 110#.

Click Next on Page 2 of the Software Upgrade wizard.

Step 8

Figure 2-15 illustrates the Upgrade Prompt that appears.

Figure 2-15 Upgrade Prom pt

Step 9

Click OK to confirm the upgrade operation.

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Step 10 The system displays Page 3 of the Software Upgrade Wizard (see Figure 2-16).

Figure 2-16 Softw are Upgrade Wizard—Page 3

Step 11 Click Select File.

The system displays the Windows Browse window (see Figure 2-17).

Figure 2-17 Use the Brow se Window to Select the Upgrade File

Step 12 Select the upgrade software (.bin file) for the selected device and click OK.

Step 13 Click Upgrade.

The system performs the upgrade and reboots the system and returns you to the wizard Main Menu. You

can upgrade another device, or perform other configuration operations.

For information about using the Setup Wizard to configure voice services from an ITSP or to configure

local SPA400 voice mail or remote ITSP voicemail, refer to Chapter 3, “Configuring V o ice Service and

V o icemail.” To configure other features, refer to Chapter 4, “Configuring S P A 9000 Features.”

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Chapter 2 Getting Started

Setting the SPA9000 Administrator Account Password

Complete the folllowing steps to change the Administrator password.

Step 1

Select Change Admin Password from the Advanced Features menu (see Figure 2-18).

The system displays the page shown in Figure 2-19.

Figure 2-18 Set SPA9000 Adm in Passw ord

Step 2

Step 3

Step 4

Type the Administrator password in the field provided.

Type the password again in the Confirm Password field and click Submit.

Click Submit to enter the configuration for the previous wizard pages.

Using the Administration Web Server

This section describes how to use the administration web server to configure the SPA9000. It includes

the following topics:

•

Connecting to the Administration Web Server, page 2-22

Administrator Account Privileges, page 2-24

Connecting to the Administration Web Server

To access the SPA9000 administration web server, perform the following steps.

Step 1

Launch a web browser on a computer connected to the SPA9000.

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Using the Administration Web Server

You access the administration web server through the Internet (WAN) interface on the SPA9000.

Determine the address of the administration web server.

Step 2

Step 3

a. Connect an analog telephone to the Phone 1 or Phone 2 port on the SPA9000.

b. Press **** on the keypad to access the IVR menu.

c. Press 110# to determine the Internet (WAN) IP address.

Direct the browser to the IP address of the SPA9000.

Figure 2-19 Adm inistration Web Server—Default Page (Basic User)

Changing between the tabs on the Voice page does not discard the unsubmitted changes. You can wait

until completing all your changes on the Voice pages before submitting them.

Note

Changing between the Router and Voice pages discards any unsubmitted changes to either page.

Step 4

Step 5

Click the Voice tab.

Click Admin Login and Advanced.

The Administrator account name for the SPA9000 is admin (with a lower-case a) and the User account

name is user. These account names cannot be changed.

The system prompts for the Administrator account password if it has been set. If prompted, type the

password provided by the ITSP and press Enter.

Step 6

To view the status information for the phones, click PBX Status.

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Chapter 2 Getting Started

Advanced Methods of Configuration

Enter the appropriate login information. Two views of the administration web server are available. Click

basic to view basic settings, or click advanced to view advanced settings.

Step 7

If you make changes on a screen, click Submit All Changes to save the changes,

Click Undo All Changes button to undo your changes. When changes are saved, the SPA9000 may

reboot.

Administrator Account Privileges

The SPA9000 supports two levels of administration privileges: Administrator and User. Both privileges

can be password protected.

Note

By default, there are no passwords assigned for either the Administrator account or the User account.

The Administrator account has the privilege to modify all the web profile parameters and can also

modify the passwords of both Administrator and User account. The User account only has the privilege

to access part of the web profile parameters. The parameters that the User account can access are

specified on the Provisioning page of the administration web server.

To directly access the Administrator account level privilege, use the following URL:

http://spa_9000_ipaddress/admin/voice

If the password has been set for the Administrator account, the browser prompts for authentication. The

User account name and the Administrator account name cannot be changed.

When browsing pages with the Administrator account privilege, you can switch to User account

privilege by clicking the User Login link.

If the User account password is set, the browser prompts for authentication when you click the User

Note

Switching between User and Administrator accounts or between basic and advanced views discards any

uncommitted changes that have already been made on the web pages.

Advanced Methods of Configuration

This section describes some advanced methods of configuration that you may find useful if you are

administering a large number of SPA9000 devices. This section includes the following topics:

•

Web Interface URLs, page 2-24

Provisioning, page 2-26

Client Auto-Configuration, page 2-27

Manual Client Configuration, page 2-28

Web Interface URLs

The SPA9000 web interface supports several functions through special URLs:

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Advanced Methods of Configuration

•

Upgrade

Reboot

Resync

•

Administrator account privilege is needed for these functions.

Upgrade URL

The Upgrade URL lets you upgrade the SPA9000 to the firmware specified by the URL, which can

identify either a TFTP or HTTP server.

Note

If the value of the <Upgrade Enable> parameter in the Provisioning page is No, you cannot upgrade the

SPA9000 even if the web page indicates otherwise.

The syntax of the Upgrade URL is as follows:

•

http://spa-ip-addr/admin/upgrade?[protocol://][server-name[:port]][/firmware-pathname]

Both HTTP and TFTP are supported for the upgrade operation.

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the

URL is used as server-name.

If no port specified, the default port of the protocol is used. (69 for TFTP or 80 for HTTP)

The firmware-pathname is typically the file name of the binary located in a directory on the TFTP or

HTTP server. If no firmware-pathname is specified, /spa.bin is assumed, as in the following example:

http://192.168.2.217/admin/upgrade?tftp://192.168.2.251/spa.bin

Resync URL

The Resync URL lets you force the SPA9000 to do a resync to a profile specified in the URL, which can

identify either a TFTP, HTTP, or HTTPS server.

Note

The SPA resyncs only when it is idle.

The syntax of the Resync URL is as follows:

http://spa-ip-addr/admin/resync?[[protocol://][server-name[:port]]/profile-pathname]

If no parameter follows /resync?, the Profile Rule setting from the Provisioning page is used.

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the

URL is used as server-name.

If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443 for HTTPS).

The profile-path is the path to the new profile with which to resync, for example:

http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg

Reboot URL

The Reboot URL lets you reboot the SPA9000.

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Note

The SPA9000 reboots only when it is idle.

The Reboot URL is http://spa-ip-addr/admin/reboot.

Provisioning

This section describes the provisioning functionality of the SPA9000. This section includes the

following topics:

•

Provisioning Capabilities, page 2-26

Configuration Profile, page 2-26

For detailed information about provisioning the SPA9000, refer to the LVS SPA Provisioning Guide.

Provisioning Capabilities

The SPA9000 provides for secure provisioning and remote upgrade. Provisioning is achieved through

configuration profiles transferred to the device via TFTP, HTTP, or HTTPS.

The SPA9000 can be configured to automatically resync its internal configuration state to a remote

profile periodically and on power up. The automatic resyncs are controlled by configuring the desired

profile URL into the device.

The SPA9000 accepts profiles in XML format, or alternatively in a proprietary binary format, which is

generated by a profile compiler tool available from Linksys. The SPA9000 supports up to 256-bit

symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial

provisioning stage), the SPA9000 can receive a profile from an encrypted channel (HTTPS with client

authentication), or it can resync to a binary profile generated by the Linksys-supplied profile compiler.

In the latter case, the profile compiler can encrypt the profile specifically for the target SPA9000, without

requiring an explicit key exchange.

Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades using HTTPS are not

supported). Remote upgrades are controlled by configuring the desired firmware image URL into the

SPA9000 via a remote profile resync.

For further information about remote provisioning refer to the LVS SPA Provisioning Guide. For further

information about certificate generation for use with an HTTPS server, contact [email protected].

Configuration Profile

The SPA9000 configuration profile can be either an XML file or a binary file with a proprietary format.

The XML file consists of a series of elements (one per configuration parameter), encapsulated within

the element tags <flat-profile> … </flat-profile>. The encapsulated elements specify values for

individual parameters. Here is an example of a valid XML profile:

<flat-profile>

<Admin_Passwd>some secret</Admin_Passwd>

<Upgrade_Enable>Yes</Upgrade_Enable>

</flat-profile>

Binary format profiles contain SPA9000 parameter values and user access permissions for the

parameters. By convention, the profile uses the extension .cfg (for example, spa2000.cfg). The Linksys

Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly

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formatted and encrypted .cfg file. The SPC tool is available from Linksys for the Win32 environment

and Linux-i386-elf environment. Requests for SPC tools compiled on other platforms are evaluated on

a case-by-case basis. Please contact your Linksys sales representative for further information about

obtaining the SPC tool.

The syntax of the plain-text file accepted by the profile compiler is a series of parameter-value pairs,

with the value in double quotes. Each parameter-value pair is followed by a semicolon. Here is an

example of a valid text source profile for input to the SPC tool:

Admin_Passwd “some secret”;

Upgrade_Enable “Yes”;

Refer to the LVS SPA Provisioning Guide for further details.

The names of parameters in XML profiles can generally be inferred from the SPA9000 configuration

Web pages, by substituting underscores (_) for spaces and other control characters. Further, to

distinguish between Lines 1, 2, 3, and 4, corresponding parameter names are augmented by the strings

_1_, _2_, _3_, and _4_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.

Parameters in the case of source text files for the SPC tool are similarly named, except that to

differentiate Line 1, 2, 3, and 4, the appended strings ([1], [2], [3], or [4]) are used. For example, the

Line 1 Proxy is named Proxy[1] in source text profiles for input to the SPC.

Client Auto-Configuration

An unprovisioned client station in the factory default state can be automatically provisioned by the

SPA9000 by following the flow chart shown in Figure 2-20.

When the SPA9000 receives a request for /cfg/init_$MA.xml, it automatically assigns the next available

user ID (extension number) to this client station. The next user ID to be assigned to a new client station

is configured using The <Next Auto User ID> parameter and is automatically incremented each time a

new number is assigned. Before assigning a new user ID, the SPA9000 also checks whether there is any

registered client station using that ID and keeps increasing the ID until an unused value is found.

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Figure 2-20 Unprovisioned Client Station Acquiring a Configuration Profile

DHCP

Yes

Request

/spa$PSN.cfg

Look For

Config Server

Option 66?

Request

/cfg/init_$MA.xml

Request

/cfg/generic.xml

To add a new IP phone to the SPA9000, connect the IP phone to the QoS switch to which the SPA9000

is connected and power on the unit.

To add a previously-used IP phone to the SPA9000, perform the following steps:

1. Upgrade the IP phone with SPA9000-compatible firmware.

2. Factory reset the unit.

3. Power cycle the unit.

4. Connect the unit to the switch.

The SPA9000 provisions only the necessary parameters to the client stations. It assumes the rest of the

parameters have appropriate values, which are either the default values or manually configured values.

For example, the SPA9000 provisions only Extension 1 on the client stations. Access the administration

web server using the Administrator account to manually configure other extensions on specific client

stations.

Manual Client Configuration

The client stations can also be manually configured with the contents of the profile served by the

SPA9000.

The following XML file is served by the SPA9000 when a client station requests /spa$PSN.cfg

<flat-profile>

<Resync_Periodic>1</Resync_Periodic>

<Profile_Rule>tftp://spa-9000-ip-address:69/cfg/init_$MA.xml</Profile_Rule>

</flat-profile>

The following XML file is served by the SPA9000 when client station requests /cfg/init_$MA.xml:

<flat-profile>

<User_ID_1_>next-available-user-id</User_ID_1_>

<Extension_1_>1</Extension_1_>

<Short_Name_1_>next-available-user-id</Short_Name_1>

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<Extension_2_>1</Extension_2_>

<Short_Name_2_>next-available-user-id</Short_Name_2>

<Extension_3_>1</Extension_3_>

<Short_Name_3_>next-available-user-id</Short_Name_3>

<Extension_4_>1</Extension_4_>

<Short_Name_4_>next-available-user-id</Short_Name_4>

<Station_Name>client-station-mac-address</Station Name>

<Resync_Periodic>1</Resync_Periodic>

<Resync_Error_Retry_Delay>10</Resync_Error_Retry_Delay>

<Profile_Rule>tftp://spa-9000-ip-address:69/cfg/generic.xml</Profile_Rule>

<Linksys_Key_System>1</Linksys_Key_System>

</flat-profile>

The following XML file is served by the SPA9000 when a client station requests /cfg/generic.xml:

<flat-profile>

<Resync_Periodic>0</Resync_Periodic>

<Resync_Error_Retry_Delay>3600</Resync_Error_Retry_Delay>

<Admin_Passwd>spa-9000-admin-passwd</Admin_Passwd>

<Password_1_>phone-ext-password</Password_1_>

<Proxy_1_>spa-9000-ip-address:proxy-listen-port</Proxy_1_>

<Voice_Mail_Server_1_>spa-9000-ip-address:proxy-listen-port </Voice_Mail_Server_1_>

<Voice_Mail_Number>vmm</Voice_Mail_Number>

<Cfwd_Busy_Dest>vm</Cfwd_Busy_Dest>

<Cfwd_No_Ans_Dest>vm</Cfwd_No_Ans_Dest>

<Multicast_Address>spa-9000-multicast-address</Multicast_Address>

<Upgrade_Rule>phone-upgrade-rule</Upgrade_Rule>

<Dial_Plan>phone-dial-plan</Dial_Plan>

<Linksys_Key_System>1</Linksys_Key_System>

<Remote_Party_ID_1_>1</Remote_Party_ID_1_>

<Time_Zone>time-zone</Time_Zone>

<Daylight_Saving_Time_Rule>daylight-saving-time</Daylight_Saving_Time_Rule>

</flat-profile>

Table 2-3 lists the variables used in these XML files.

Table 2-3 Variables Used in XML Configuration Files

Variable

Description

spa-9000-ip-address

proxy-listen-port

IP address of the SPA9000 SIP Proxy.

Port at which the SPA9000 SIP Proxy is listening. This value is

configured in <Proxy Listen Port>.

client-station-mac-address

This is the MAC address of the client station who is requesting the

profile /cfg/init_$MA.xml (in other words, the $MA portion of the

requested filename).

next-available-user-id

Phone-upgrade-rule

The current value of <Next Auto User ID>.

Upgrade rule to be used by the client stations. This value is

configured in <Phone Upgrade Rule>.

Phone-dial-plan

Dial plan to be used by the client stations. This value is configured

in <Phone Dial Plan>.

time-zone

<Time Zone> value that is configured on the SPA9000.

daylight-saving-time

<Daylight Saving Time Rule> value that is configured on the

SPA9000.

phone-ext-password

<Phone Ext Password> value configured on the SPA9000.

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Table 2-3 Variables Used in XML Configuration Files

Variable

Description

spa-9000-admin-passwd

spa-9000-multicast-address

<Admin Passwd> value configured on the SPA9000.

<Multicast Address> value configured on the SPA9000.

Client stations download spa$PSN.cfg and init_$MA.xml only once for initial configuration. However,

they download generic.xml on every reboot. Therefore, parameters manually configured on the client

station that overlap with the contents of generic.xml are overwritten with the SPA9000-supplied values.

The list of parameters included in generic.xml are thus purposely kept to a very small set.

Client Registration

All client stations served by the SPA9000 must register to the SPA9000, which does not allow a station

to make calls unless it is registered. If the client station is configured with Station Name, it should

include a P-STATION-NAME header in the REGISTER request. Following is an example where User-A

has been assigned a primary extension of 5031.

Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK-8865c41e

From: “User-A” <sip:[email protected]:6060>;tag=a76a3e1dfc6045cdo0

To: “User-A” <sip:[email protected]:6060>

Call-ID: [email protected]

CSeq: 1 REGISTER

Max-Forwards: 70

Contact: “User-A” <sip:[email protected]:5060>;expires=3600

User-Agent: Sipura/SPA841-3.1.4(a0714sec)

P-Station-Name: User-A

Content-Length: 0

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

In this example, User-A shares a line appearance with User-B, whose primary extension is 5041.

Extension 2 on the User-A station must then be set up the same way as Extension 1 (User-A primary

extension), but with the <Shared User ID> parameter set to 5041. The User-A station then performs a

third-party registration for Extension 2, as shown below. Note that the TO header <user-id> parameter

is the User-B primary extension.

Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK-25c8108c

From: “User-A” <sip:[email protected]:6060>;tag=3c43d094a9424bo1

To: “User-A” <sip:[email protected]:6060>

Call-ID: [email protected]

CSeq: 1 REGISTER

Max-Forwards: 70

Contact: “User-A” <sip:[email protected]:5061>;expires=3600

User-Agent: Sipura/SPA841-3.1.4(a0714sec)

P-Station-Name: 000e08daf417

Content-Length: 0

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

The 200 reply sent by the SPA9000 to the client station REGISTER request includes a DATE header that

the client station can use to synchronize with its local real-time clock. The time served in the DATE

header is the local time (as opposed to GMT). There is thus no need to configure an NTP server or a time

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zone on the client stations. This assumes that the SPA9000 can maintain the real-time clock based on an

NTP server or a DATE header supplied by the ITSP. Following is an example of a 200 response to

SIP/2.0 200 OK

To: “User-A” <sip:[email protected]:6060>;tag=41a7-0

From: “User-A” <sip:[email protected]:6060>;tag=8d0bd416dc8a7ec2o0

Call-ID: [email protected]

CSeq: 1 REGISTER

Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK-e62fe987

Contact: sip:[email protected]:5060;expires=3600

Content-Length: 0

Date: Mon, 18 Jul 2005 14:39:40 PST

Troubleshooting and Configuration FAQ

This section provides solutions to problems that may occur during the installation and

operation of the SPA9000. If you can't find an answer here, check the Linksys website at

www.linksys.com.

1. The SPA9000 did not automatically assign an extension number to the Linksys IP Phone, and the

phone Ext LED is yellow instead of green.

There are two solutions to the common causes of this problem.

First solution:

A. Connect an analog phone to the SPA9000 and access the IVR by pressing ****.

B. Press 180# to hear the multicast address and port assigned to the SPA9000.

C. If the problem is with a SPA901, perform these steps to verify and correct the multicast address:

•

Use option 180# on the IVR to determine the multicast address of the phone that is not getting

configured.

•

If the addresses or ports are different, press 181# to set the address and port of the phone to

match the SPA9000. Use a * to separate each set of digits in the IP address and before the port

number.

D. If the problem is with any SPA900 Series phone with an LCD display, perform these steps to

verify and correct the multicast address:

•

Press the Menu button (to the right of the 6) and scroll down to Network and press Select.

•

Scroll to the bottom of the Network list (you can get there quickly by pressing the up arrow)

and view the Multicast Address.

•

If the address and port does not match the SPA9000, press Edit, and enter the correct address

and port. Then press OK.

E. Restart the SPA900 Series phone.

Second solution:

A. Open the web browser on the administration computer.

B. Enter http://ipaddress/admin/router/status.

C. Write down the Current IP of the SPA9000. (This is the Internet IP address.)

D. Access the phone administration web server at the following URL:

http://phone_ip_address/

To determine the IP address of the phone, press the Menu button and select Network from the

LCD menu. For a SPA901 phone, use IVR option 110#.

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E. Make sure the configured proxy server on the phone matches the SPA9000 IP address. (Refer to the

phone documentation for details.)

2. The IP phone can make internal calls to other IP phones and analog phones but it cannot make

external calls.

Check to see whether the SPA9000 line is registered. Follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://ipaddress/admin/voice/advanced.

C. On the Voice - Info screen, check to see if the Line 1 Status indicates that the Registration State says,

“Registered.”

D. If it is not registered, then verify that the User ID, Proxy, and Password supplied by your Internet

Telephony Service Provider (ITSP) are valid (these settings are configured on the Line page).

3. I made a call from an outside line, and I did not hear a ring tone after I entered the extension number.

First, try again and make sure you entered the extension number correctly. If you still do not hear a ring

tone, then follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://ipaddress/admin/voice/status.

C. On the PBX Status screen, make sure the IP phone for that extension number is registered.

4. I made a call from an outside line, and the Auto-Attendant says, “Not a valid extension, please try

again.” However, I can make outgoing calls from the IP phone with that extension number.

Follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://ipaddress/admin/voice/advanced.

C. Click the SIP tab.

D. On the Voice - SIP screen, add the extension number to the dial plan.

5. When an outside line calls the SPA9000, it rings one time and then goes to the Auto-Attendant.

By default, if no one answers the call after 12 seconds, then the call goes to the Auto-Attendant. To

change this setting, follow these steps:

A. Open the web browser on the administration computer.

B. Enter http://ipaddress/admin/voice/advanced.

C. Click the SIP tab.

D. On the Voice - SIP screen, change the appropriate Answer Delay setting (DayTime, NightTime, or

Weekends/Holidays).

6. How can I change greetings for the Auto-Attendant?

Connect an analog phone to the SPA9000 FXS port (Phone 1 or Phone 2) and use the Interactive Voice

Response Menu to record or change greetings. Refer to the“Using the Interactive V o ice Response

Interface, page 2-8” for instructions.

7. I want to use a different computer to access the administration web server. I entered http://ipaddress,

but this address did not work.

Any computer connected to your router should use the Internet (WAN) IP address of the SPA9000. Use

the Interactive Voice Response Menu to find out the SPA9000 Internet IP address. Follow these steps:

A. Try pinging the IP address you are using. If you don’t get a reply, make sure the device is working,

connected, and that you have the correct address.

B. To determine the correct IP address of the administration web server:

a. Use a telephone connected to the Phone 1 port of the SPA9000.

b. Press **** (in other words, press the star key four times).

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c. Wait until you hear “Linksys configuration menu. Please enter the option followed by the #

(pound) key or hang up to exit.”

d. Press 110#.

e. You hear the IP address assigned to the SPA9000 Internet (external) interface. Write it down.

C. To enable access to the administration web server:

a. Press 7932#.

b. Press 1 to enable WAN access to the administration web server.

8. I’m trying to access the SPA9000 administration web server, but I do not see the login screen. Instead,

I see a screen saying, “404 Forbidden.”

If you are using Windows Explorer, perform the following steps until you see the administration web

server login screen (Netscape Navigator requires similar steps):

A. Click File. Make sure Work Offline is NOT checked.

B. Press CTRL + F5. This is a hard refresh, which forces Windows Explorer to load new webpages, not

cached ones.

C. Click Tools. Click Internet Options. Click the Security tab. Click the Default level button. Make

sure the security level is Medium or lower. Then click the OK button.

9. How do I save my current SPA configuration?

Currently, the only way is to do HTTPGET from an HTTP client, from which you get the entire

HTML page. Alternatively, from your browser you can select File > Save as > HTML from any of

the administration web server pages. Do this in Admin, Advanced mode.

This saves all the tabs into one HTML file. This HTML file is helpful to provide to our support team

when you have a problem or technical question.

10.How do I debug my SPA? Is there a syslog?

SPA sends out debug information via syslog to a syslog server. The ports can be configured (by

default the port is 514).

A. Make sure you do not have firewall running on your PC that could block port 514.

B. On the administration web server System tab, set <Debug Server> as the IP address and port

number of your syslog server. Note that this address has to be reachable from the SPA9000).

C. Also, set <Debug level> to 3.

You do not need to change the value of the <syslog server> parameter.

D. To capture SIP signaling messages, under the Line tab, set <SIP Debug Option> to Full.

The file output is syslog.<portnum>.log (for the default port setting, syslog.514.log)

11. How do I view the status of my VoIP devices when I can’t access the administration web server on

the network?

A. Connect a regular Ethernet cable to the Ethernet (LAN) port of the SPA9000.

Note

The Ethernet port on the SPA9000 is not intended for use to connect the device to the

network.

B. Direct your web browser to the following URL:

http://192.168.0.1/admin/router/status

C. If you do not get a response, ping the interface to verify that it is working, or use IVR option

#210 to verify the address on the LAN interface.

D. Click PBX Status on the Line tab.

12.How do I access the SPA9000 if I forget my password?

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By default, the Admin account has no password. If a password is set for this account and you do not

know what it is, you have to reset the device to its factory defaults. There are two options for

resetting the device:

Option 877778 resets the unit to the configuration shipped from the ITSP and erases any

configuration completed locally by the user.

Option 73738 resets the unit to the configuration shipped by Linksys and erases local configuration

any provisioning configured by the ITSP.

To reset the SPA9000 to the factory defaults, perform the following steps:

A. Connect an analog phone to the SPA9000 and access the IVR by pressing ****.

B. Press 877778# to reset the unit to the defaults as it shipped from the ITSP or press 73738# to reset the

unit to the defaults as it shipped from Linksys.

C. Press 1 to confirm the operation. Press * to cancel the operation.

D. Login to the unit using the Admin account without a password and reconfigure the unit.

13.My SPA9000 is behind a NAT device or firewall and I’m unable to make a call or I’m only

receiving a one-way connection. What should I do?

A. Configure your router to port forward “TCP port 80" to the ip address currently being used by

SPA. If you do this often, we suggest that you use static IP address for the SPA, instead of

DHCP. (For help with port forwarding, consult your router documentation)

B. On the Line tab of the administration web server, change the value of <Nat Mapping Enable> to

yes. On the SIP tab; change <Substitute VIA Addr> to yes, and <EXT IP> to the IP address of

your router.

C. Make sure you are not blocking the UDP PORT 5060,5061 and port for UDP packets in the

range of 16384-16482. Also, disable “SPI” if this feature is provided by your firewall. Identify

the SIP server to which the SPA9000 is registering, if it supports NAT, using the <Outbound

Proxy> parameter.

D. Adjust the NAT keepalive variables as follows:

E. Check the following settings on the administration web server:

F. Add a STUN server to allow traversal of UDP packets through the NAT device. On the SIP tab of the

administration web server, set <STUN Enable> to yes, and enter the IP address of the STUN server

in <STUN Server>.

STUN (Simple Traversal of UDP through NATs) is a protocol defined by RFC 3489, that allows a

client behind a NAT device to find out its public address, the type of NAT it is behind, and the port

associated on the Internet connection with a particular local port. This information is used to set up

UDP communication between two hosts that are both behind NAT routers. Open source STUN

software can be obtained at the following website:

http://www.voip-info.org/wiki-Open+Source+VOIP+Software

Note

STUN does not work with a symmetric NAT router. Enable debug through syslog (see FAQ#10),

and set <STUN Test Enable> to yes. The messages indicate whether you have symmetric NAT

or not.

14. I manually entered the proxy address into my SPA941 but it doesn't register with the SPA9000.

A. Make sure to have the SPA9000 IP address and the corresponding port, the default port is 6060 <

Proxy Listen Port>. For example, enter 192.168.0.1:6060 in the Proxy location of the SPA941.

15.Why can’t I register the FXS ports to an outside proxy?

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A. The FXS ports are registered directly with the SPA9000; no options exists for it to connect to

another proxy.

16. Why does the SPA941 reboot when rebooting the SPA9000?

A. When booting, the SPA9000 sends a multicast message to request all client stations to gracefully

reboot. This allows the client stations to redo DHCP discovery, download latest phone parameters,

and also register with the SPA9000.

17.Why do only four SPA900 Series clients register?

A. The SPA9000 ships with a four-user license. To support more clients, purchase a 16-user license

and install it by entering the license key in the < License Key> parameter on the Provisioning

page of the web server administration server. To obtain a license key contact

[email protected].

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C H A P T E R

Configuring Voice Service and Voicemail

This chapter describes how to configure voice use voicemail with the SPA9000 and includes the

following sections:

•

Using the Wizard to Configure ITSP V o ice Services and V o icemail, page 3-1

Completing the V o icemail Configuration, page 3-6

Configuring the SPA400 for PSTN Connectivity or V o icemail, page 3-8

S P A 400 V o icemail Options, page 3-20

Configuring a SPA IP Phone for V o ice Mail Service, page 3-18

Managing V o icemail, page 3-20

Using the Wizard to Configure ITSP Voice Services and

Voicemail

After using the Setup Wizard to complete the initial configuration of the SPA9000 (see the “Using the

Wizard for Initial Configuration” section on page 2-13) the system displays the Main Menu (Figure 3-1).

Complete the following steps to set up voice services or voicemail with an ITSP.

Note

If you use the Setup Wizard after using the administration web server to change your configuration, you

may lose any changes to the factory default configuration that were made using the administration

server.

Step 1

If you have closed the Setup Wizard, perform these steps to display the Main Menu.

1. Start the Wizard by double-clicking the executable file.

2. Click Next until you get past the introductory screens.

3. Select I have configured the SPA9000 using the Wizard before and click Next again.

4. On the screen that appears, enter the IP address of the SPA9000 and a password if it is set

The system displays the Main Menu.

To configure voice services or voicemail with an ITSP, complete the following steps.

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Chapter 3 Configuring Voice Service and Voicemail

Using the Wizard to Configure ITSP Voice Services and Voicemail

Figure 3-1 Setup Wizard—Main Menu

Step 2

Select Configure SPA9000 from the Main Menu.

The system displays the page shown in Figure 3-2.

Figure 3-2 Configuring the Connection Type

Step 3

Step 4

Select Using an ITSP to configure ITSP voice servers or voicemail.

If you are using both ITSP and the SPA400 for voicemail or PSTN connectivity, select Using both ITSP

and PSTN and refer to the “Configuring the S P A 400 for PSTN Connectivity or V o icemail” section on

page 3-8 for instructions about using the Wizard to configure the SPA400.

Click Next.

The system displays the page shown in Figure 3-3.

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Chapter 3 Configuring Voice Service and Voicemail

Using the Wizard to Configure ITSP Voice Services and Voicemail

Figure 3-3 Configuring the ITSP Voice Service

Step 5

Configure the options on this page as directed by the Internet Telephony Service Provider (ITSP).

The following describes the purpose of each parameter on this page.

•

Proxy—Enter the fully qualified domain name of the SIP proxy server for all outbound requests.

This modifies the <Proxy> parameter on the NAT Settings section of the Line tab on the

administration web server.

Note

If you use the administration web server to change the configuration of a parameter, using the

Wizard again may overwrite your changes. To avoid this, either rerun the Wizard to change your

configuration, or do not use the Wizard after using the administration web server.

•

Outbound Proxy—Enter the fully qualified domain name of a SIP outbound proxy server where all

outbound requests are sent as the first hop. This modifies the <Outbound Proxy> parameter.

Enable NAT Keep Alive—If you enter yes, the configured <NAT Keep Alive Msg> is sent

periodically every <NAT Keep Alive Intvl> seconds. This modifies the <NAT Keep Alive Enable>

parameter.

•

NAT Keep Alive Interval—Enter an integer to determine the interval in seconds between

NAT-mapping keep alive messages. The default is 15. This modifies the <NAT Keep Alive

Enable> parameter.

NAT Keep Alive Message—Enter the contents of the keep-alive message to be sent to a given

destination periodically to maintain the current NAT-mapping. This can be an empty string. If value

is $NOTIFY, a NOTIFY message is sent as keep alive. If value is $REGISTER, a REGISTER

message without Contact is sent. This modifies the <NAT Keep Alive Msg> parameter.

•

NAT Keep Alive Destination—Enter the fully qualified domain name of the destination to send NAT

keep alive messages. If the value is $PROXY, it is sent to the current proxy or outbound proxy. This

modifies the <NAT Keep Alive Dest> parameter.

•

User ID—Enter the value provided by your ITSP.

Password—Enter the value provided by your ITSP.

Display Name—This is the string that is used for displaying Caller ID information.

Auth ID—Enter the SIP authentication ID if this is required by your ITSP.

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Chapter 3 Configuring Voice Service and Voicemail

Using the Wizard to Configure ITSP Voice Services and Voicemail

Step 6

Click Next.

The system displays the page shown in Figure 3-4.

Figure 3-4 Configure SPA9000 Voicem ail Server

Step 7

Select the appropriate option on Page 3 of the wizard.

•

Option 1: My ITSP provides multiple voicemail accounts

Option 2: I have a separate voicemail server which provides a voicemail account for each extension

Option 3: I don’t have multiple voicemail accounts from either my ITSP or a separate voicemail

service provider

Step 8

Click Next and configure the appropriate options on the version of Page 5 that appears.

Figure 3-5 Configuring the Service Provider Voicem ail Server

If you selected Option 1 for Step 7 (see Figure 3-4), the system displays the version of Page 5 shown in

Figure 3-5.

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Chapter 3 Configuring Voice Service and Voicemail

Using the Wizard to Configure ITSP Voice Services and Voicemail

Note

Contact the ITSP/VMSP for the correct configuration parameters if a URL is not used for the

voicemail server link.

If you selected Option 2 for Step 7 (see Figure 3-4), the system displays the version of Page 5 shown in

Figure 3-6.

Figure 3-6 Configuring a Third-Party Voicem ail Server

Complete the fields on Page 5 as directed by the ITSP.

The following describes the purpose of each parameter on both versions of Page 5.

•

Voicemail Proxy—Enter the fully qualified domain name of a SIP outbound proxy server where all

outbound requests are sent as the first hop. This modifies the <Outbound Proxy> parameter.

Outbound Proxy—Enter the fully qualified domain name of a SIP outbound proxy server where all

outbound requests are sent as the first hop.

Mailbox Subscribe URL—URL that should receive the SUBSCRIBE messages, so the SPA9000

receives voicemail status notification for all mailboxes on this line. This parameter allows $USER

and $PROXY macros, such as $USER@$PROXY.

•

Mailbox Deposit URL—URL that the SPA9000 contacts when clients and external callers need to

deposit voicemail in any of the mailboxes on this line.

Mailbox Manage URL—URL that the IP telephony contacts when it needs to check voicemail for

any of the mailboxes on this line.

•

UserID—Enter the value provided by your ITSP.

Password—Enter the value provided by your ITSP.

Display Name—Enter the value provided by your ITSP.

Auth ID—Enter the SIP authentication ID if this is required by your ITSP.

Step 9

After completing Page 5, click Next.

The system displays the page shown in Figure 3-7 on page 3-6.

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Chapter 3 Configuring Voice Service and Voicemail

Completing the Voicemail Configuration

This section is required to complete the voicemail configuration regardless of whether you use ITSP

voicemail, a third-party voicemail server, or the SPA400 local voicemail option.

Step 1

On Page 6 of the wizard, verify or complete the configuration of each line extension and enable the

Mailbox for each extension.

Figure 3-7 Configure the SPA9000 Internal Phone Extensions

Note

When you check the mailbox, the extension uses the extension number for the mailbox ID

(MBID), which is the correct setting.

Step 2

After completing Page 6, click Next.

The system displays the page shown in Figure 3-8.

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Chapter 3 Configuring Voice Service and Voicemail

Completing the Voicemail Configuration

Figure 3-8 Configuring the SPA9000 Call Routing Rule

Step 3

Step 4

Use the options on Page 7 to identify the following options as required at your site:

•

Ring the Auto-Attendant after waiting a specific number of seconds after receiving a call.

Ring another extension before ringing the Auto-Attendant.

Ring the Auto-Attendant immediately.

After completing Page 7, click Next.

The system displays the page shown in Figure 3-9.

Figure 3-9 Localization

You can use the Localization option on the Advanced Features menu to change your selection after

completing this part of the Wizard (see the “Using the Wizard for Localization” section on page 4-7).

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Step 5

Select the correct value for your location from each of the pull-down selection lists.

The system displays a final confirmation page.

.Click Submit to confirm the configuration changes you have made.

Step 6

Configuring the SPA400 for PSTN Connectivity or Voicemail

This section describes how to configure the SPA400 for PSTN connectivity or voicemail. It includes the

following topics:

•

Understanding How the S P A 400 Interacts with the SPA9000, page 3-8

Using the SPA9000 Setup Wizard to Configure the SPA400, page 3-8

Using the Administration Web Server to Configure the S P A 400, page 3-11

Configuring SPA400 V o icemail Accounts, page 3-17

Note

When using the Setup Wizard to configure the SPA400, disable other network cards on the PC where

you are running the Setup Wizard except for the card that is connected to the SPA9000 through the local

switch. Otherwise, the Setup Wizard may fail to properly locate the SPA400.

Understanding How the SPA400 Interacts with the SPA9000

The SPA400 features the ability to connect up to four standard analog telephones lines to a Linksys Voice

SPA9000 (LVS) VoIP network and includes the additional benefit of a voicemail application. A total of

four SPA400 devices can be configured per SPA9000, using up to 16 analog phone lines and

automatically routing calls to and from your existing PSTN telephone service.

Use the SPA400 for access to the PSTN by connecting its FXO ports to analog lines provisioned by the

local carrier. There are specific Line settings on the SPA9000 that are configured to recognize a SPA400

as a service enabling calls from an IP phone on a LVS network to be routed to the PSTN and vice versa.

The SPA400 sets up and tears down calls between the PSTN and IP phones, offering seamless telephone

service with Linksys VoIP equipment.

Using the SPA9000 Setup Wizard to Configure the SPA400

It is recommended that you use the SPA9000 Setup Wizard to install and configure the SPA400. Version

1.1.0.3 (or higher) of the SPA9000 Setup Wizard supports the SPA400 for the initial or an existing LVS

installation. Connect the SPA400 to the network using its Ethernet port before running the SPA9000

Setup Wizard. The latest version of the SPA9000 Setup Wizard can be downloaded from

http://www.linksys.com.

Note

If you use the Setup Wizard to configure the SPA9000 after using the administration web server, you

may lose any changes to the factory default configuration made with the administration server.

To configure voice services or voicemail with the SPA400, complete the following steps.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Step 1

If you have closed the Setup Wizard, start it again.

Select I have configured the SPA9000 using the Wizard before on the first page of the Wizard. Then

on the second page, enter the IP address of the SPA9000 and the password and click Next and the system

displays the Main Menu. To determine the IP address of the SPA9000, use IVR option 110.

After using the Setup Wizard to complete the initial configuration of the SPA9000, as described in the

“Using the Wizard for Initial Configuration” section on page 2-13, the system displays the Main Menu

(Figure 3-1).

Step 2

Select Configure SPA400 from the Main Menu.

The system displays the page shown in Figure 3-10.

Figure 3-10 Configure the SPA400—Page 1

Step 3

Step 4

Complete the configuration of Page 1 as follows:

•

Select the SPA400 MAC address from the pull-down selection list.

Type the Administrator account password, if this has been set.

Click Next.

The system displays the page shown in Figure 3-11.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Figure 3-11 Configure the SPA400—Page 2

Step 5

Select the way you are using SPA400 and click Next.

The system displays the page shown in Figure 3-12.

Figure 3-12 SPA400 Disconnection Tone Localization

Step 6

Select the appropriate country from the pull-down selection list and click Next.

The choices in the current release are US and Australia.

The system displays the page shown in Figure 3-13.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Figure 3-13 SPA400 Disconnection Tone Settings

Step 7

Review the settings on this window and click OK.

The system displays the page shown in Figure 3-14.

Figure 3-14 SPA400 Subm it Confirm ation

Step 8

Click Submit to save and activate your changes.

To complete the SPA9000 voicemail configuration, perform the steps described in the “Completing the

V o icemail Configuration” section on page 3-6.

Using the Administration Web Server to Configure the SPA400

This section describes how to use the SPA400 and SPA9000 administration web servers. It includes the

following topics:

•

Accessing the Administration Web Server, page 3-12

Configuring the SPA400 to Connect to the SPA9000, page 3-12

Configuring the SPA9000 to Register the S P A 400, page 3-15

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Accessing the Administration Web Server

The SPA400 and the SPA9000 have corresponding parameters that must be properly configured for a

successful installation. If you do not use the SPA Setup Wizard, you can complete the configuration

using the administration web server provided by each device.

Note

If you use the Setup Wizard to configure the SPA400 after using the administration web server, you may

lose any changes to the SPA9000 factory default configuration made with the administration server.

To access the SPA400 administration web server, direct your browser to the IP address of the SPA400.

Running the SPA9000 Setup Wizard simplifies installation and automatically detects the IP address of

the SPA400. If you are using DHCP, to determine the IP address of the SPA400, view the DHCP clients

table for the DHCP server that assigned the address to the SPA400.

After entering the IP address of the SPA400 into the browser, a password request page appears. The

administrator account name for the SPA400 is Admin (with a capital A) and with the default

configuration, the password field should be left blank (no password). Click the OK button to access the

SPA400 administration web server.

Note

Requiring a password to access the SPA400 prevents unauthorized changes to the settings of the device.

To assign a password, select the Administration tab on the administration web server and then click

Management.

Configuring the SPA400 to Connect to the SPA9000

The following steps demonstrate how to configure a SPA400 on an LVS network with a SPA9000. The

example configuration assumes that a SPA9000 is configured to access the PSTN only, and therefore the

SPA400 is configured in the Line 1 tab on the SPA9000.

If a VoIP service provider is also defined, a SPA400 can be configured in any of the available Line tabs

(1–4) on the SPA9000 administration web server.

Step 1

Step 2

From the SPA400 administration web server, click SPA9000 Interface on the Setup tab.

Enter the user ID of the SPA9000 in the User ID field.

The User ID setting must be identical to the User ID configured in the associated field in the Line # field

on the SPA9000.

The Discover Automatically radio button enables the SPA400 to learn the IP address and the UDP port

of the SPA9000 from the SIP registration packets sent by the SPA9000.

Note

A static IP address is recommended for the SPA400, but the SPA9000 may be assigned its IP

address by the DHCP server on the local router or directly from the ISP over the WAN link.

Step 3

If you are using a static IP address for the SPA9000, enter this address and the UDP port to enable the

SPA400 to listen for the SIP registration request from the SPA9000.

If the SPA9000 obtains an IP address from a DHCP server, leave the Discover Automatically option

enabled to avoid registration failure because of a possible change in the SPA9000 IP address or

expiration of the DHCP lease.

Figure 3-15 and Figure 3-16 show the administration web server for the SPA400.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Figure 3-15 SPA400 Adm inistration Web Server—Setup/ SPA9000 Interface Page

Figure 3-16 shows the administration web server for the SPA9000. In the configuration example shown

in these figures, the SPA400 is listed as the Line 1 service, permitting PSTN access for the SPA9000.

The User ID field is configured with SPA9000 for the User ID, which is identical the value configured

on the SPA400.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Figure 3-16 SPA9000 Configuration Utility—Voice/ Line 1 Page

Step 4

Set the Registration and Subscribe parameters.

After the SPA400 reboots, an ITSP caller forwarded into the SPA400 voice mail does not hear a voice

prompt unless the SPA9000 is also rebooted. To ensure that the SPA9000 reboots each time the SPA400

is rebooted, change the values of the following parameters on the Proxy and Registration section of the

SIP tab.

•

Mailbox Subscribe Expires: change from 2147483647 to 30

The SPA9000 then detects when the SPA400 reboots automatically re-registers and resubscribes.

Note

The default value for these parameters has been changed to 30 in Release 5.1.7 and later, so the

default values do not have to be changed.

Step 5

Click Save Settings at the bottom of the page and then click Administration > Reboot > Restart

System.

The SPA400 device reboots.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Configuring the SPA9000 to Register the SPA400

This section describes how to configure three settings on the SPA9000 that are required to register the

SPA400 with the SPA9000.

Step 1

Step 2

Select the appropriate Line tab from the SPA9000 administration web server .

On the Line page for the line assigned to the SPA400, make sure the following fields match the settings

entered in the SPA400 configuration:

•

SIP settings—SIP port = 5060

This is the default SIP port. Only change this value if a different port is configured on the SPA400

for some reason.

Subscriber information—User ID = SPA400_User_ID

Replace SPA400_User_ID with the value entered on the User ID field of the SPA9000 Interface on

the Setup tab of the SPA400 In the example, this value is SPA9000.

Proxy and registration—Proxy = SPA400 IP Address

Replace SPA400_IP_Address with the actual IP address of the SPA400. In the example, this is

192.168.1.103.

Step 3

Click the Submit All Changes button on the SPA9000 administration web server.

The SPA9000 device reboots.

The SPA9000 is now configured to access the PSTN through the SPA400.

Configuring the SPA400 Voicemail Services

The SPA400 includes a USB voicemail module for users (or extensions) configured on a SPA9000. The

USB flash drive that ships with the SPA400 provides a total of 128 MB of storage for voicemail prompts

and mailbox messages. The maximum length of an individual message is 180 seconds. Mailbox

messages require 480KB per minute of recording time, so the flash disk can contain a total of more than

3 hours of mailbox messages.

When you receive the SPA400, you can back up the voicemail prompts and other files on the USB drive

to a secure location on a PC or server on your network. You can then copy these files to a larger

compatible USB drive, in case you need more voicemail storage. To determine if a USB drive is

compatible, after copying the files to it, insert it into the USB port on the SPA400. View the status of the

<<>> parameter on the SPA400 administration web server.

The integrated voicemail application server supports 32 configurable voicemail accounts enabling users

to leave and play back voice messages. (Although a SPA9000 can be configured with up to 4 SPA400

devices, only 1 SPA400 can be configured with the voicemail application).

The SPA9000 can be configured with up to four SPA400 devices, the voice mail service can either

concentrate with one SPA400 or distribute among the four SPA400. However, an IP Phone can only

subscribe to one mailbox from one SPA400..

To configure the SPA400 to register with the SPA9000, complete the following steps.

Step 1

From the SPA400 administration web server, click Voicemail Server on the Setup tab (see Figure 3-17).

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Figure 3-17 SPA400 Adm inistration Web Server—Setup/ Voicem ail Server Page

Step 2

In the Server Port field, enter the UDP port over which the Voicemail Server listens for signaling

between the SPA400 and the SPA9000.

In Figure 3-17, the value of this field is 5090.

Note

This port number must be different from the UDP port configured in the SPA9000 Interface

menu.

Step 3

Enter the User ID that the SPA9000 uses to register the SPA400 voicemail server.

This setting must be identical to the User ID configured in the SPA400 SPA9000 Interface page. In the

example, this value is SPA9000.

Step 4

Step 5

Step 6

Step 7

In the SPA9000 subscriber ID field, enter the Subscriber ID that the SPA9000 uses to subscribe to the

SPA400 voicemail server for obtaining notification.

In the Mailbox deposit number field, enter the associated telephone number that the SPA9000 uses to

deposit voicemail.

In the Mailbox manage number field, enter the associated telephone number that the SPA9000 uses to

access voicemail.

Click Save Settings at the bottom of the page and then the Restart button on the SPA400 administration

web server.

The SPA400 device reboots.

Configuring the SPA9000 Voicemail Settings

You must match the parameters entered in the SPA400 Voicemail Server fields in the SPA9000

administration web server under the associated Line # used for the SPA400 voicemail server.

The example assumes the SPA9000 is configured to use Line 1. The configuration and installation of

the SPA400 and the integrated voicemail application server can be configured on the same SPA9000

Line tab.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring the SPA400 for PSTN Connectivity or Voicemail

Step 1

Step 2

Three settings (URL) must be configured to enable SPA9000 users to subscribe, access, deposit, and

manage their voicemail services. These settings are entered in their respective fields on the SPA9000 in

the following format:

<service_id#>@<spa400_ip_address>:<port>

Where service_id# and port must match the setting entered in the SPA400 Voicemail Server menu.

From the appropriate SPA9000 Line tab, make sure that the following fields match the settings entered

in the SPA400 Voicemail Server page.

In the current example, the settings would be as follows:

•

Subscriber information—User ID = SPA9000

SPA9000 User ID

Proxy and registration—Mailbox deposit URL= [email protected]:5090

mailbox deposit number

Proxy and registration—Mailbox manage URL= [email protected]:5090

mailbox manage number

Proxy and registration—Mailbox subscribe URL= [email protected]:5090

SPA9000 subscriber ID

Step 3

Click the Submit All Changes button on the SPA9000 administration web server.

The SPA9000 device reboots.

The SPA400 and the SPA9000 are now configured to use the integrated voicemail application server.

Configuring SPA400 Voicemail Accounts

The SPA400 supports up to 32 voicemail accounts allowing LVS users to receive and playback voicemail

messages. Creating a voicemail account for an LVS user requires entering a Mailbox ID in the

configuration of the SPA IP Phone and then associating it to a SPA400 Voicemail User ID.

A Voicemail User ID is defined as a valid SPA IP Phone extension that has been configured in the

SPA9000. To add or change voicemail User IDs from the SPA400 administration web server, complete

the following steps.

Step 1

From the SPA400 administration web server, click the Setup tab and select Voicemail Users (see

Figure 3-18).

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Chapter 3 Configuring Voice Service and Voicemail

Configuring a SPA IP Phone for Voice Mail Service

Figure 3-18 SPA400 Adm inistration Web Server—Setup/ Voicem ail Users Page

If the installation and configuration of the SPA400 was initially performed using the SPA9000

Configuration Wizard, each extension configured for voicemail is populated in the SPA400 Voicemail

Users page.

By default, a user voicemail account is active with the password to access the voicemail account being

identical to their extension (that is, ext.=100 password=100).

Step 2

Verify or modify the settings on the either the SPA400 or SPA900 pages so that each extension of each

SPA900 Series IP phone matches the corresponding settings on the SPA400 Voicemail Users page.

Step 3

Step 4

If you make any changes, click Save Settings at the bottom of the SPA400 page.

If you made any changes to the SPA400 page, click Restart button on the SPA400 page.

The SPA400 device reboots if you save changes.

Configuring a SPA IP Phone for Voice Mail Service

Each valid SPA IP Phone extension configured in the SPA9000 may be configured for voicemail service.

The SPA IP Phone configuration page (Figure 3-19) can be accessed by either clicking on the PBX

Status link in the upper right location of the SPA9000 configuration utility or by directing the browser

to the IP address of the SPA900 Series phone.

Step 1

On the SPA IP Phone administration web server, click the Ext 1 tab.

Figure 3-19 illustrates the page for Extension 1 on the SPA900 Series IP phone administration web

server.

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Chapter 3 Configuring Voice Service and Voicemail

Configuring a SPA IP Phone for Voice Mail Service

Figure 3-19 SPA900 Series IP Phone Adm inistration Web Server—Ext 1 Page

Step 2

Enter the voicemail setting in the Mailbox ID field in the following format:

<line#><User ID>

Where <line#> refers to the SPA9000 Line (1–4) of the SPA400 configuration and <User ID> must

match the setting entered in the SPA400 Voicemail Users menu.

For the current example, the setting is as follows:

Call feature settings—Mailbox ID = 1100

The IP address reflected in the Voicemail Server field on the SPA IP Phone Ext 1 page (Figure 3-19) is

correctly configured as the IP address of the SPA9000.

The SPA9000 acts as a proxy to the SPA400 voicemail server.

Click the Submit All Changes button on the SPA900 Series administration web server.

The SPA900 Series phone reboots.

Step 3

The SPA900 Series phone has now been configured to access the voicemail service.

Note

Refer to the SPA IP Phone Voicemail Quick Reference Card for details about using the voicemail service.

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Chapter 3 Configuring Voice Service and Voicemail

SPA400 Voicemail Options

The following list the options that the voicemail user enters on the telephone keypad for each option in

the voicemail system provided by the SPA400.

Note

The VM button on SPA900 Series IP phones provides direct access to the SPA400 voicemail system.

Press * to return to the Main Menu.

1 Access new message(s)

3 Access Advanced options

1 send a reply

3 heard the message envenlop (time and caller)

5 send a VM message

7 undelete the message

* return to the Main Menu

5 repeat current message

3 advanced option

5 repeat

7 delete

8 forward

9 save

* help

2 Change folders; change to which folder

0 New messages

1 Old message

2 Work messages

3 Family messages

4 Friend messages

# Cancel

3 Advanced options

5 Send a VM message

* Return to main menu

6 Next message

7 Delete/undelete

8 Forward to another user

9 Save message: which folder

New message

Old message

Work messages

Family messages

Friends message

# Cancel

Mail box options

Record unavailable message

Record busy message

Record name

Change password

* Help

# Exit

Managing Voicemail

On the SPA9000, accessing a voicemail server to check or deposit voicemail is similar to calling an

external number, or being call forwarded to an external number. Each line interface can use a different

voicemail server.

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Chapter 3 Configuring Voice Service and Voicemail

Managing Voicemail

There are three groups of voicemail operations:

Depositing voicemail

Managing voicemail from a client station or from an external number

Subscription to voicemail notification and receiving voicemail notification

•

SPA9000 assumes that a voicemail account can include more than one mailbox ID (MBID). The

voicemail account is defined with a user-ID, which can be the same as the line interface user-ID.

Three parameters must be configured on the SPA9000 for each line to support these operations:

<Mailbox Deposit URL>, <Mailbox Manager URL>, and <Mailbox Subscribe URL>.

Note

The mailbox ID should be set to the extension number.

Voicemail service may be offered by a service provider different from the ITSP. For example, you can

configure Line 1, 2, and 3 with accounts on an ITSP, but configure Line 4 an account with a different

Internet voicemail service provider (IVMSP). The SPA9000 can be configured to bridge calls between

the ITSP and the IVMSP when necessary (when depositing or checking voicemail by an external caller)

using the <VMSP Bridge> parameter on each line interface.

How Voicemail Works

When a user checks voicemail from a client station, the SPA9000 sends an INVITE on its behalf to the

configured <Mailbox Manage URL>. For example:

INVITE sip:mailbox-manage-url SIP/2.0

Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK-171eb6b5

From: <sip:[email protected]>;tag=300704dd2590d20bo2;ref=5031;mbid=53371

To: <sip:mailbox-manage-url>

Call-ID: [email protected]

CSeq: 101 INVITE

Max-Forwards: 70

Contact: <sip:[email protected]:5060> Ö

Here the client station is at extension 5031 and the mailbox ID is 53371. The voicemail server should

then prompt the caller to enter a PIN and access the voicemail features for the given mailbox.

A user should also be able to call an external number explicitly from anywhere to retrieve voicemails.

When the voicemail server receives such a call, it should prompt the caller to enter the mailbox ID and

then the PIN number. If the user-ID of the voicemail account cannot be uniquely identified from the

mailbox ID or from the called number, the server must first prompt the user to enter the user-ID before

proceeding.

Note

If an EXT-To-DID mapping exists for the calling extension on the line interface, the user-ID and display

name fields of the FROM header are replaced by the mapped DID number and the display name assigned

to the phone, respectively. In this case, a DIVERSION header similar to the REFERRED-BY header is

also included.

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Chapter 3 Configuring Voice Service and Voicemail

Managing Voicemail

Checking Voicemail from an External Number

The Internet voicemail service provider (IVMSP) may have an external number for their subscribers to

call to check/manage their mailboxes. If this is not available, you may do one of the following:

•

(a) assign a DID number (from the ITSP)

(b) create a virtual extension for this purpose.

For method (a), specify a rule in the <Contact List> for the dedicated DID number. For the DID number

18005551000, this would look like the following example:

...|...|18005551000:vmm3|...

The syntax vmm<n> tells the SPA9000 to forward calls dialed to 18005551000 to the voicemail

management URL on Line <n>. Because no particular mailbox is specified in this example, the caller is

prompted to enter the voicemail number after the voicemail server answers. You may also specify a

particular mailbox ID with this syntax (for example, vmm31234).

Method (b) is used in conjunction with the Auto-Attendant. For example, you can define a single-digit

extension 7 to map to the voicemail management URL on Line 4, by adding a rule to <AA Dial Plan 1>

or <AA Dial Plan 2> depending on which one you are using. The rule would look like the following:

<7:vmm4>

Again, you may also specify a particular mailbox in this syntax. For information about configuring the

Auto-Attendant, refer to Chapter 5, “Configuring the LVS Auto-Attendant.”

Depositing Voicemail

An external caller can be triggered to deposit voicemail into a mailbox by sending it a REFER request

during a call with <Mailbox Deposit URL> indicated in the REFER-TO header and the mailbox ID in

the REFERRED-BY header. The caller device then sends INVITE to the <Mailbox Deposit URL> as

shown in the example below:

INVITE sip:mailbox-deposit-url SIP/2.0

Via: SIP/2.0/UDP 24.35.36.111;branch=z9hG4bK-29752ae9

From: “External Caller” <sip:[email protected]>;tag=b99e21414928473o2

To: <sip:mailbox-deposit-url>

Call-ID: [email protected]

CSeq: 101 INVITE

Contact: <sip:[email protected]>

Referred-By: <sip:[email protected]>;ref=5041;mbid=7675

In the last example, the caller is directed to deposit voicemail in the mailbox ID 7675 on the voicemail

account 37683101. It further indicates that the directing station is at the internal extension 5041. Note

that it is assumed that the caller device (or the ITSP) is faithfully relayed over the REFERRED-BY

header.

Note

The mailbox ID must be set to the extension number.

The INVITE sent for an internal caller to deposit voicemail is similar, except that the FROM header also

includes a reference parameter, as shown in the following example:

INVITE sip:mailbox-deposit-url SIP/2.0

Via: SIP/2.0/UDP 172.12.244.56;branch=z9hG4bK-29752ae9

From: Line 1 <sip:[email protected]>;tag=b99e21414928473o2;ref=5031

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Chapter 3 Configuring Voice Service and Voicemail

Managing Voicemail

To: <sip:mailbox-deposit-url>

Call-ID: [email protected]

CSeq: 101 INVITE

Max-Forwards: 70

Contact: <sip:[email protected]>

Referred-By: <sip:[email protected]>;mbid=7675

In the above example, the station wanting to deposit voicemail is at extension 5031; the mailbox ID is

7675 on the account 37683101 (for the station at extension 5041). Note that the referrer is the same as

the caller in the last INVITE. In other words, this INVITE is self-triggered by the SPA9000.

If an EXT-To-DID mapping exists for the calling extension on the line interface, the user-ID and display

name fields of the FROM header is replaced by, respectively, the mapped DID number and the display

name assigned to the phone. Furthermore, if an EXT-To-DID mapping exists for the called phone, the

user-ID field of the REFERRED-BY header is also replaced by the mapped DID number for the called

extension.

Note

For information about how to add an EXT-to-DID mapping, see the <Contact List> parameter in

Chapter 6, “SPA9000 Field Reference”

Subscribing to Voicemail Notification

The SPA9000 sends a one-time SUBSCRIBE for the message-summary event package for each line with

a valid <Mailbox Subscribe URL>. The SUBSCRIBE implies subscription for the status of all the

mailboxes associated with the voicemail account used-ID; it does not include any mailbox ID in the

request. Following is an example:

SUBSCRIBE sip:mailbox-subscribe-url SIP/2.0

Via: SIP/2.0/UDP 172.16.22.23:5062;branch=z9hG4bK-44f9d0f0

From: Line 3 <sip:[email protected]>;tag=ac6013983cce7526

To: <sip:mailbox-subscribe-url>

Call-ID: [email protected]

CSeq: 63017

SUBSCRIBE Max-Forwards: 70

Contact: <sip:[email protected]:5062>

Expires: 2147483647

Event: message-summary

User-Agent: Sipura/SPA9000-3.2.2

Content-Length: 0

Note that the Expires value is purposely set to a very large number so the subscription does not expire

and there is no need to renew the subscription. The voicemail server is expected to send a NOTIFY

immediately upon receiving this SUBSCRIBE message for each mailbox on this account. The

Request-URI of the NOTIFY should reference the CONTACT header of the corresponding

SUBSCRIBE, but the user-ID in the To header should be the mailbox ID. The following example shows

a NOTIFY for the mailbox ID 5031 on the account 14089991003:

NOTIFY sip:[email protected]:5062 SIP/2.0

Via: SIP/2.0/UDP 178.178.221.230;branch=z9hG4bK-44f9d0f0

From: <sip:[email protected]>;tag=ab789

To: <sip:[email protected]:5062>;tag=ac6013983cce7526

Call-ID: [email protected]

CSeq: 537

NOTIFY Expires: 2147483647

Event: message-summary User-Agent: ITSP/Voicemail-Server

Content-Length: 0

Messages-Waiting: yes

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Chapter 3 Configuring Voice Service and Voicemail

Managing Voicemail

Voice-Message: 2/8 (0/2)

Note

Note that SPA9000 does not require the NOTIFY to be sent within the same subscription dialog. That

is, it accepts the NOTIFY even without a TO-tag or a matching Call-ID as the original SUBSCRIBE.

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C H A P T E R

Configuring SPA9000 Features

This chapter describes how to perform the basic configuration for enabling the SPA9000 features. It also

provides detailed explanations of how the system performs each operation, and the parameters

controlling system behavior. This chapter includes the following sections:

•

Using the Wizard to Configure SPA9000 V o ice Features, page 4-1

Using Dial Plans, page 4-9

Basic Call Management, page 4-13

Managing Call Forwarding, page 4-17

Managing Call Pickup, page 4-25

Multicast and Group Paging, page 4-27

Music On Hold, page 4-28

Enhancements in Release 5.1, page 4-32

Using the Wizard to Configure SPA9000 Voice Features

You can use the SPA9000 Setup Wizard to enable or configure many of the main voice features. This section,

which describes the options available, includes the following topics:

•

Configuring Client Stations, page 4-2

Configuring Client Extensions, page 4-4

Configuring Shared Extensions, page 4-5

Configuring a Call Hunt Group, page 4-6

Using the Wizard for Localization, page 4-7

For information about configuring the Auto-Attendant, refer to Chapter 5, “Configuring the LVS

Auto-Attendant.”

Note

If you use the Setup Wizard after using the administration web server to change the SPA9000

configuration, you may lose any changes to the factory default configuration that were made using the

administration server web pages.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Configuring Client Stations

To configure the extensions for client stations connected to the SPA9000, complete the following steps.

Step 1

Select Configure Client Stations on the Setup Wizard Main Menu.

After using the Setup Wizard to complete the initial configuration of the SPA9000, as described in the

“Using the Wizard for Initial Configuration” section on page 2-13, the system displays the Main Menu

(Figure 4-1).

If you have closed the Setup Wizard, perform these steps to display the Main Menu:

1. Start the Wizard by double-clicking the executable file.

2. Click Next until you get past the introductory screens.

3. Select I have configured the SPA9000 using the Wizard before and click Next again.

4. On the screen that appears, enter the IP address of the SPA9000 and a password if it is set

The system displays the Main Menu (see Figure 4-1).

Figure 4-1 SPA Setup Wizard—Main Menu

Step 2

Click Next.

The system displays Page 1 of the wizard. As explained on this page, complete the following steps to

begin configuring each phone:

1. Power on the IP phone.

2. If the phone has been used previously, reset the phone to its factory defaults.

•

For the SPA901, use the IVR by pressing ****, press 73738#, and finally press 1#

For SPA92x, 942, or 962, press Menu, Navigate down to Factory Reset, and click Select.

3. Make sure the phone is connected to the switch.

4. Click Next on Page 1.

The system displays Page 2.

Step 3

Determine the firmware version.

•

For the SPA901, use the IVR by pressing *** and press 150#.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

•

For SPA92x, 94x, or 925, press Menu on the IP phone and Navigate down to Product Info >

Software Version.

Step 4

If the firmware version is lower than 4.1.12, click Upgrade.

The system displays Page 3, shown in Figure 4-2.

Figure 4-2 Client Station Setup Wizard—Page 3

Step 5

Select the MAC address or station name of the IP phone you are configuring and click Next.

The system displays Page 4, shown in Figure 4-3.

Figure 4-3 Client Station Setup Wizard—Page 3

Step 6

Step 7

Identify the station name, select an extension number for each available extension, and click Next.

Click Finish on Page 5, and if you want to configure additional clients, select the required option on the

final page.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Configuring Client Extensions

To use the Setup Wizard to modify the configuration of the client extensions on the SPA9000, complete

the following steps.

Step 1

To update the configuration of client stations, select Client Extensions from the Advanced Feature

Menu (see Figure 4-4).

Figure 4-4 Advanced Feature Menu

The system displays the screen shown in Figure 4-5.

Figure 4-5 Configuring Client Extensions

Step 2

Enter or modify the extension and user name that you want to assign to each line.

Use the Phone 1 and Phone 2 fields to identify the devices connected to the FXS ports.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Step 3

Click Submit to make the changes.

Configuring Shared Extensions

Use this option to enable the shared line appearance (SLA) feature, which allows multiple stations to

share an extension number and manage a call as a group. For further information about this feature, refer

to the “Using Shared Line Appearance” section on page 4-22.

To use the Wizard to enable shared extensions, complete the following steps.

Step 1

Select Client Shared Extensions from the Advanced Feature Menu (see Figure 4-6).

Figure 4-6 Client Shared Extensions Option

The system displays the page shown in Figure 4-7.

Figure 4-7 Configuring Internal Shared Extensions

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Step 2

Configure the lines that you want to share and click Submit.

Configuring a Call Hunt Group

A hunt group is a number of extensions that ring when a number is dialed. The call is answered by the

first extension that picks up the call. Extensions can be alerted simultaneously or sequentially. See the

description of the <Hunt Groups> and the <Contact List> parameters about how to specify a hunting

rule. Note that <Contact List> is a per line interface parameter and is applicable to that line interface

only. For information about using the administration web server to configure hunt groups, see the “Using

Call Hunt Groups” section on page 4-19.

To use the Setup Wizard to configure a hunt group, complete the following steps.

Step 1

Select Hunt Group from the Advanced Feature Menu (see Figure 4-8).

Figure 4-8 Hunt Group Option

The system displays the page shown in Figure 4-9.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Figure 4-9 Configure Hunt Groups

Step 2

Click Add Hunt Group to add a group.

Click Edit to change an existing group, or click Delete to remove a group.

When you click Add Hunt Group, the system displays the page shown in Figure 4-10.

Figure 4-10 Configure Hunt Group

Step 3

Step 4

Identify the hunt group and click OK.

When the next screen reappears, the new hunt group is added.

Click Submit to create the hunt group.

Using the Wizard for Localization

To use the wizard for setting localization settings for the SPA9000, complete the following steps.

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Chapter 4 Configuring SPA9000 Features

Using the Wizard to Configure SPA9000 Voice Features

Step 1

Select Configure Advanced Features from the Main Menu.

The system displays the Advanced Feature Menu (see Figure 4-11)

Figure 4-11 Advanced Features Menu

Step 2

Select Localization from the Advanced Feature Menu.

The system displays the Localization page (see Figure 4-12)

Figure 4-12 Localization Wizard—Page 1

Step 3

Step 4

Step 5

Select the correct value for your location from each of the pull-down selection lists.

To resync the time with the local administration PC, click Resync to PC time.

Click Submit.

The system enables the selected options and returns to the Advanced Features menu.

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Chapter 4 Configuring SPA9000 Features

Using Dial Plans

This section describes the dial plan features of the SPA9000. It also explains how to configure and write

scripts and provides a reference describing the use of each dial plan parameter. For information about

dial plans for the Auto-Attendant, refer to Chapter 5, “Configuring the L V S Auto-Attendant.”

Configuring Dial Plans

The SPA9000 allows each line to be configured with a distinct dial plan. The dial plan specifies how to

interpret digit sequences dialed by the user, and how to convert those sequences into an outbound dial

string.

The SPA9000 syntax for the dial plan closely resembles the corresponding syntax specified by MGCP

and MEGACO. Some extensions are added that are useful in an end-point.

Dial Plan Digit Sequences

The plans contain a series of digit sequences, separated by the | character. The collection of sequences

is enclosed in parentheses.

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The

matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the

set of candidates diminishes until only one or none are valid.

Any one of a set of terminating events triggers the SPA9000 to either accept the user-dialed sequence

and transmit it to initiate a call, or else to reject it as invalid. The terminating events are as follows:

•

No candidate sequences remain—The number is rejected.

Only one candidate sequence remains, and it has been matched completely—The number is

accepted and transmitted after any transformations indicated by the dial plan, unless the sequence

is barred by the dial plan, in which case the number is rejected.

•

A timeout occurs—The digit sequence is accepted and transmitted as dialed if incomplete, or

transformed as per the dial plan if complete.

An explicit “send” (user presses the # key)—The digit sequence is accepted and transmitted as

dialed if incomplete, or transformed as according to the dial plan if complete.

The time-out duration depends on the matching state. If no candidate sequences are as yet complete (as

dialed), the Interdigit_Long_Timeout applies. If a candidate sequence is complete, but there exists one

or more incomplete candidates, then the <Interdigit_Short_Timeout> applies.

Table 4-1 describes the entries to use when programming the dial plan.

Table 4-1 Dial Plan Entries

Dial Plan Entry

Function

*xx

Allows arbitrary 2-digit star code

Allows x11 sequences (for example, 311, 411, 611, 911)

Dials operator

[3469]11

Dials international operator

Dials US local number

[2-9]xxxxxx

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Chapter 4 Configuring SPA9000 Features

Using Dial Plans

Table 4-1 Dial Plan Entries

Dial Plan Entry

1xxx[2-9]xxxxxx

xxxxxxxxxx.

Function

Dials US 1 + 10-digit long distance number

Dials all other numbers, including international long distance

Dial Plan Rules

This section describes the rules that apply to configuring and interpreting dial plans.

Note

White space is ignored, but may be used for readability.

Digit Sequence Syntax

Each digit sequence within the dial plan consists of a series of elements, which are individually matched

to the keys pressed by the user. Elements can be one of the following:

•

Individual keys 0, 1, 2 . . . 9, *, #.

The letter x matches any one numeric digit (0 .. 9)

A subset of keys within brackets (allows ranges): for example, [389] means 3 or 8 or 9)

–

Numeric ranges (n-n) are allowed within the brackets: for example, [2-9] means any digit from

2 through 9)

–

Ranges can be combined with other keys: e.g. [235-8*] means 2 or 3 or 5 or 6 or 7 or 8 or *.

Element Repetition

Any element can be repeated zero or more times by appending a period (.) to the element. Thus, “01.”

matches “0”, “01”, “011”, “0111”, … and so on.

Sub-sequence Substitution

A sub-sequence of keys (possibly empty) can be automatically replaced with a different sub-sequence

using an angle bracket notation: < dialed-subsequence : transmitted-subsequence >. So, for example,

“<8:1650>xxxxxxx” would match “85551212” and transmit “16505551212”.

Inter-sequence Tones

Number Barring

An “outside line” dial tone can be generated within a sequence by appending a comma (,) between digits.

Thus, the sequence “9, 1xxxxxxxxxx” sounds an “outside line” dial tone after the user presses 9, until

the 1 is pressed.

A sequence can be barred (rejected) by placing a ! character at the end of the sequence. Thus,

“1900xxxxxxx!” automatically rejects all 900 area code numbers from being dialed.

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Chapter 4 Configuring SPA9000 Features

Using Dial Plans

Interdigit Timer Master Override

The long and short interdigit timers can be changed in the dial plan (affecting a specific line) by

preceding the entire plan with the following syntax:

•

Long interdigit timer: L : delay-value ,

Short interdigit timer: S : delay-value ,

Thus, “L:8,( . . . )” would set the interdigit long timeout to 8 seconds for the line associated with this

dial plan. And, “L:8,S:4,( . . . )” would override both the long and the short time-out values.

Local Timer Overrides

The long and short time-out values can be changed for a particular sequence starting at a particular point

in the sequence. The syntax for long timer override is: “L delay-value<space>“. Note the terminating

space character. The specified delay-value is measured in seconds. Similarly, to change the short timer

override, use: S delay-value<space>.

Pause

A sequence may require an explicit pause of some duration before continuing to dial digits, in order for

the sequence to match. The syntax for this is similar to the timer override syntax: P delay-value <space>.

The delay-value is measured in seconds.

This syntax allows for the implementation of Hot-Line and Warm-Line services. To achieve this, one

sequence in the plan must start with a pause, with a 0 delay for a Hot Line, and a non-zero delay for a

Warm Line.

Implicit Sequences

The SPA9000 implicitly appends the vertical code sequences entered in the administration web server

Regional parameter settings to the end of the dial plan for both Line 1 and Line 2. Likewise, if

<Enable_IP_Dialing> is enabled, then IP dialing is also accepted on the associated line.

Dial Plan Examples

The following dial plan accepts only US-style 1 + area-code + local-number, with no restrictions on the

area code and number:

( 1 xxx xxxxxxx )

The following also allows 7-digit US-style dialing, and automatically inserts a 1 + 212 (local area code)

in the transmitted number.

( 1 xxx xxxxxxx | <:1212> xxxxxxx )

For an office environment, the following plan requires a user to dial 8 as a prefix for local calls and 9 as

a prefix for long distance. In either case, an “outside line” tone is played after the initial 8 or 9, and

neither prefix is transmitted when initiating the call.

( <9,:> 1 xxx xxxxxxx | <8,:1212> xxxxxxx )

The following allows only placing international calls (011 call), with an arbitrary number of digits past

a required 5 digit minimum, and also allows calling an international call operator (00). In addition, it

lengthens the default short interdigit timeout to 4 seconds.

S:4, ( 00 | 011 xxxxx x. )

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Chapter 4 Configuring SPA9000 Features

Using Dial Plans

The following allows only US-style 1 + area-code + local-number, but disallows area codes and local

numbers starting with 0 or 1. It also allows 411, 911, and operator calls (0).

( 0 | [49]11 | 1 [2-9]xx [2-9]xxxxxx )

The following allows US-style long distance, but blocks 9xx area codes:

( 1 [2-8]xx [2-9]xxxxxx )

The following allows arbitrary long distance dialing, but explicitly blocks the 947 area code.

( 1 947 xxxxxxx ! | 1 xxx xxxxxxx )

The following implements a hot line phone, which automatically calls 1 212 5551234.

( S0 <:12125551234> )

The following provides a warm line to a local office operator (1000) after five seconds, unless a

four-digit extension is dialed by the user.

( P5 <:1000> | xxxx )

Dial Plan Timers

The dial plan functionality is regulated by the following configurable parameters:

•

Interdigit_Long_Timer

Interdigit_Short_Timer

Dial_Plan ([1] and [2])

Interdigit Long Timer

ParName

Default

Interdigit_Long_Timer

The <Interdigit_Long_Timer> specifies the default maximum time (in seconds) allowed between dialed

digits, when no candidate digit sequence is as yet complete (see the discussion of the Dial_Plan

parameter for an explanation of candidate digit sequences).

Interdigit Short Timer

ParName

Interdigit_Short_Timer

Default

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Chapter 4 Configuring SPA9000 Features

Basic Call Management

The <Interdigit_Short_Timer> specifies the default maximum time (in seconds) allowed between dialed

digits, when at least one candidate digit sequence is complete as dialed (see the following discussion of

Dial_Plan parameters for an explanation of candidate digit sequences).

Dial Plans

ParName

Default

Dial_Plan[n] for Each Line n

(*xx | [3469]11 | 0 | 00 | <:1408>[2-9]xxxxxx |

1[2-9]xx[2-9]xxxxxx | 011x. )

The Dial_Plan parameters contain the actual dial plan scripts for each line n, where n is a number from

1 to 4.

Basic Call Management

•

Receiving External Phone Calls, page 4-13

Calling Between Client Stations, page 4-13

Client Stations Calling an External Number, page 4-14

External Users Calling the S P A 9000, page 4-15

Supporting Multiple DID Numbers Per Line Interface, page 4-16

Receiving External Phone Calls

To receive external phone calls from the Internet, the SPA9000 must be configured with the Direct Inward

Dialing (DID) number assigned by the ITSP. Usually this is the same as the user ID, but it can be different.

You must also identify the IP phones that ring when an outside caller calls each DID number. The default is

aa, which stands for Auto-Attendant, an automated system that picks up external calls and plays

pre-recorded voice messages. If you want only the Auto-Attendant to receive a call, keep the default setting.

When the Auto-Attendant receives a call, it prompts the caller to dial the appropriate extension.

To identify specific IP phones to ring when the DID number is called, use the <Contact List> parameter,

described in Chapter 6, “SPA9000 Field Reference.”

Calling Between Client Stations

To make an internal call from a client station, a user dials the extension number of the target client

station. The initial INVITE message is sent to the SIP Proxy. One of the following can happen:

•

No client station currently registered with the called number—The SPA9000 replies 404 to the

calling station and terminates the call.

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Basic Call Management

•

Only one client station currently registered with the called number—The SPA9000 proxies the

INVITE to the target client station.

Multiple client stations currently registered with the called number—The SPA9000 replies 300

along with a list of contacts to the calling station and terminates the call. The calling station is

expected to fork the INVITE to all the stations in the list of contacts inserted by the SPA9000 in the

300 response.

The SPA9000 participates only in the initial call setup. After the call is set up, client stations exchange

SIP messages and RTP packets directly without involving the SPA9000 (unless the SPA9000 is

configured as a media proxy).

A client station can also call any of the hunt groups defined in the <Hunt Groups> parameter like a

normal extension. The SPA9000 performs the hunting on behalf on the calling station until the call has

been answered by a member of the hunt group, or until the hunting expires or fails.

Client Stations Calling an External Number

The proxy determines that a called number is external if it does not match any currently registered user,

and it then matches it to one of the call routing rules specified in <Call Routing Rule>.

Note

If the called number is “vm” or “vmm”, the SPA9000 handles the call differently (it deposits or retrieves

voicemail, respectively).

For calling an external number, the SPA9000 selects one of the line interfaces (1–4) with which to make

the external call. This choice is made as follows:

1. The order in which to select a line to make the call depends on the group to which the caller belongs.

A caller can belong to one of five groups: Group 0, Group 1, Group 2, Group 3, or Group 4. A caller

belongs to Group x if its user ID (that is, its extension number) matches one of the patterns specified

in <Group x User ID>, where x = 1,2,3,4. If it matches multiple groups, then the smallest group

number is assumed. If it matches none of the groups, it is considered as belonging to Group 0, which

is also known as the default group.

Each group has a corresponding <Group x Line> parameter for x =1,2,3,4, whereas <Default Group

Line> corresponds to Group 0. The contents of <Group x Line> and <Default Group Line>

determine the order of the lines to try for callers belonging to the corresponding group. For example,

if the caller belongs to Group 2 and <Group 2 Line> = 4,1,3, the SPA9000 tries to use Line 4, then

Line 1, then Line 3 to make the call.

2. The SPA9000 attempts to pick the line in the order determined in Step 1. A line is selected if:

•

It is enabled in the configuration and functioning.

It matches the routing rule for the called number.

It has capacity to take more calls.

Its <Dial Plan> allows the called number.

3. SPA9000 goes through all the allowed lines determined in Step 1 until a line is selected.

A <Call Routing Rule> is a special dial plan with a collection of rules of the form

<:Lw,x,y,z>number-pattern, where w, x, y, z are the lines that are allowed to make the call if the called

number matches the given number-pattern. For example:

<:L1,2>9xx.|<:L3>011852xx.|<:L4>*1800xxxxxxx

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Basic Call Management

This example specifies the following:

•

Lines 1 and 2 are allowed when the called number starts with 9

Line 3 is allowed if the called number starts with 011852

Line 4 is allowed if the number starts with *1800 followed by seven more digits.

One or more client station patterns can be specified in <Group x Line>, with each pattern separated by

a comma. Each pattern may contain the “*” and “?” wildcards, or “%xx” escaped characters. For

example:

<Group x Line> = 51*,56%2a*,577?

In this example, any number starts with 51 or 56*, and any four-digit number starting with 577 belongs

to Group x.

Each line interface has a finite call capacity (maximum number of simultaneous calls allowed on the

line), which is statically configured in the line <Call Capacity> parameter. When the maximum call

capacity is reached, the SPA9000 does not allocate any more calls to that line interface.

Each line interface behaves as a Back-To-Back User Agent (B2BUA).

Note the following:

•

The INVITE sent to the ITSP carries a “ref” URI parameter in the FROM header that specifies which

is the extension. Below is an example where the 5031 is the extension that originates the call to the

external number 37683103.

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK-9ee9fbdd

From: PBX Line 1

<sip:[email protected]>;tag=4625e6e97e37072ao2;ref=5031

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 101 INVITE

Max-Forwards: 70

Contact: PBX Line 1 <sip:[email protected]:5060>...

•

The default <Phone Dial Plan> for provisioning the client stations is:

(9,[3469]11S0|9,<:1408>[2-9]xxxxxx|9,<:1>[2-9]xxxxxxxxxS0|9,1[2-9]xxxxxxxxxS0|

9,011xx.|9,xx.|[1-8]xxx)

This dial plan specifies to dial 9 to get an outside line. The <Dial Plan> of each line interface, on

the other hand, is set to (<9:>xx.) which strips the leading digit 9 from the number dialed by the

client station.

External Users Calling the SPA9000

When the SPA9000 receives an external incoming call to its line interfaces (Line 1, 2, 3, or 4), it consults

the line <Contact List> to identify the client stations to alert. The <Contact List> parameter may contain

one or more number patterns, with each pattern separated by a comma. Each pattern may contain the “*”

and “?” wildcard or “%xx” escaped characters.

For example:

Line 1 <Contact List> = 531?,aa.

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Basic Call Management

In this example, the SPA9000 alerts all client stations with a four-digit user ID that starts with 531, and

also the internal Auto-Attendant when it receives an incoming call on Line 1. Furthermore, if multiple

contacts are currently registered with a user-id matching the <Contact List> parameter, the SPA9000

alerts all the registered contacts.

Note

The SPA9000 alerts all registered client stations if <Contact List> = *.

Supporting Multiple DID Numbers Per Line Interface

It is possible for the ITSP to identify the local client stations to which an external incoming call should

be routed. This can be done in several ways. Linksys recommends that the ITSP indicate this information

in the TO header of the incoming INVITE while the request-URI is still addressed to the line interface

user-id.

For example, suppose the Line 1 account has the main number 4085553000, which comes with a block

of ten DID numbers: 4085553000–4084443009. When the ITSP sends the INVITE to the SPA9000 Line

1, it can indicate the DID number in the TO header user-id field, such as:

INVITE sip:[email protected] SIP/2.0

To: <sip:[email protected]>

…

Alternatively, the DID number can be indicated as a parameter in the TO header with a configurable

parameter name, such as “didn”. For example:

INVITE sip:[email protected] SIP/2.0

To: <sip:[email protected]>;didn=4089993003

…

Identify the field to use for the DID number and the parameter name in the SPA9000 <SIP DIDN Field>

and <SIP DIDN Param Name> parameters. In the first example, <SIP DIDN Field> = “TO UserId” and

<SIP DIDN Param Name> are ignored; in the second example, <SIP DIDN Field> = “TO Param” and

<SIP DIDN Param Name> = “didn.”

You can configure the SPA9000 to route calls to a client station based on DID numbers embedded in the

INVITE message by using the full syntax of <Contact List>, which allows multiple rules with each rule

separated by a “|”. The syntax is as follows:

<Contact List> = rule[|rule[|rule[...]]]

where:

rule = [did:]ext[,ext[,ext[…]]]

did = Embedded DID number. If the DID is not specified, the rule applies to any DID number

ext = Client extension number pattern; may contain * or ? wildcard or %xx escaped characters

The maximum length of the <Contact List> parameter is 383 characters.

For the previous example, you could specify the following:

<Contact List> = 4089993000:aa|4089993001:3001|4089993002:3002|…|4089993009:3009

Thus, any external caller calling the main number is answered by the Auto-Attendant, while the other 9

DID numbers are assigned to dedicated private extensions.

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Chapter 4 Configuring SPA9000 Features

Managing Call Forwarding

This section describes the way the SPA9000 manages call forwarding. It includes the following topics:

•

How Call Forwarding Works

Using Call Hunt Groups, page 4-19

Client Station Blind Transfers External Caller To DID/Hunt Group

Using Shared Line Appearance

Note

Some improvements in call forwarding have been introduced in Release 5.1 that are described in the

“Bridge Mode” section on page 4-32.

How Call Forwarding Works

SPA9000 supports the call forward scenarios listed in Table 4-2.

Table 4-2 Call Forw ard Scenarios

Called Party Caller Forward Target

Remarks

Client station Client station Client station

Client station Client station External

Proxy only; no direct involvement

Very similar to calling external number

Client station External

Client station

External

ACKs the 302 from called party. Then

INVITEs the target. NOTE: If the original

INVITE is forked to more than one client

station, call forward is not performed.

ACKs the 302 from called party. Sends 200 to

caller, then blind REFERs caller to target.

NOTE: If the original INVITE is forked to

more than one client station, call forward is

not performed

External

Client station External

ACKs the 302 from called party, then

INVITEs the target

SPA9000 supports call forward by client station:

•

unconditionally

when busy

on no answer

Note the following:

•

SPA9000 does not distinguish cases (1) and (2), which only makes a difference at the client station

side.

•

Case (3) is simply a delayed version of (1) or (2) where the 302 response from the called client

station is received after a 180 response (that is, ringing has started at the called station). For cases

(1) or (2), the 302 response is typically received before any 180 response.

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Managing Call Forwarding

•

When the call forward destination is “vm,” the SPA9000 forwards the call to the URL specified in

<Mailbox Deposit URL>.

Following is an example where the station at extension 5034 forwarded the external caller to an external

number 37683104 (the line interface dial plan strips the leading 9):

•

Client station replies 302 to SPA9000 (Line 2 private side)

SIP/2.0 302 Moved Temporarily

To: <sip:[email protected]:6060>;tag=e7d2be737eb6d185i0

From: External Caller <sip:[email protected]>;tag=8eb8e35678762306o3

Call-ID: [email protected]: 101 INVITE

Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK-923a6716

Contact: <sip:[email protected]:6060>

Diversion: “User-C” <sip:[email protected]:6060>;reason=unconditional

Server: Sipura/SPA841-3.1.4(a0714sec)

Content-Length: 0

•

SPA9000 (Line 2 Public Side) Replies 200 to External Caller followed by REFER:

SIP/2.0 200 OK

To: <sip:[email protected]>;tag=b47365fa7b1ae0b6i3;ref=5034

From: External Caller <sip:[email protected]>;tag=156b2582302c90cao0

Call-ID: [email protected]

CSeq: 102 INVITE

Via: SIP/2.0/UDP matrix.sipura.com:5060;branch=f87169c5-5dd80aa8-a55271cf-e6750b2d-1

Via: SIP/2.0/UDP 192.168.2.228:5060;received=192.168.2.228;branch=z9hG4bK-ef32c1f0

Contact: PBX Line 2 <sip:[email protected]:5061>

Content-Length: 103

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura

Content-Type: application/sdp

v=0

o=- 74 166 IN IP4 192.168.0.1

s=-

c=IN IP4 0.0.0.0

t=0 0

m=audio 12345 RTP/AVP 0

a=sendonly

...

REFER sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.205:5061;branch=z9hG4bK-50d452e

From: <sip:[email protected]>;tag=b47365fa7b1ae0b6i3;ref=5034

To: External Caller <sip:[email protected]>;tag=156b2582302c90cao0

Referred-By: PBX Line 2 <sip:[email protected]>;ref=5034

Call-ID: [email protected]

CSeq: 101 REFER

Max-Forwards: 70

Contact: PBX Line 2 <sip:[email protected]:5061>

Refer-To: <sip:[email protected]>

Content-Length: 0

...

When an incoming call from the ITSP is forked to multiple client stations, the SPA9000 does not honor

the 3xx response returned by any of the client stations. If you wish the call to be forwarded to voicemail

or another client station when it is not picked up, you can specify the optional “cfwd=target” syntax in

<Contact List> so that the call is forwarded to the target number if not answered.

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Managing Call Forwarding

Note

You can also specify a final forwarding destination if no station answers the incoming call and the total

hunt time has expired. Typically, the forwarding destination is a voicemailbox. See the description of the

<Contact List> parameter on how to specify a call forward destination.

Using Call Hunt Groups

This section describes how hunt groups are configured using the parameters available on the

administration web server. It includes the following topics:

•

Overview, page 4-19

Configuring a Hunt Group Rule, page 4-19

Using the Contact List Parameter, page 4-20

Overview

A hunt group is a number of extensions that share responsibility for answering a call. The extensions in

the hunt group can be alerted simultaneously or sequentially. The first extension to pick up the call

establishes a private connection to the caller.

Configuring a Hunt Group Rule

To configure a hunt group, you can use the SPA9000 Setup Wizard (see the “Configuring a Call Hunt

Group” section on page 4-6). However, after using the administration web server to change the SPA9000

configuration, these changes may be lost if you subsequently use the Setup Wizard.

To configure hunt groups using the administration web server, use the <Hunt Groups> parameter on the

PBX Parameters section. Use the <Contact List> parameter on the Subscriber Information section of a

specific Line tab to identify the extensions assigned to each hunt group.

The default value of <Contact List> is aa, for Auto-Attendant. You can configure the Auto-Attendant to

notify a caller on the associated line of the extensions assigned to each hunt group. For example, “To

contact technical support, dial extension 6001. To contact sales, dial extension 6002.” For information

about configuring Auto-Attendant voice prompts, see the “Using the IVR to Record Auto-Attendant

Prompts” section on page 5-2.

The general syntax for a hunt group rule is as follows:

ext-in:name=hunt-groupname,hunt-list,hunt=algo;ring-interval;max,cfwd=ext-fwd

Where:

•

ext-in is the extension to which the hunt group is assigned.

hunt-group is a list of comma separated extensions that form the hunt group.

algo is the method of determining the next number in the group to ring. The possible values are:

•

ne—The next number in the hunt group list is selected.

ra—A number is selected pseudo-randomly from the list.

re—Starts from the beginning of the list.

•

ring-interval is the number of seconds to ring one number in the group before trying the next.

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•

max is the number of times to cycle through the entire hunt group if the value is smaller than

ring-interval. If max is larger than ring-interval, it is interpreted as the total number of seconds to

ring all lines in the hunt group before forwarding or rejecting the call. If max is 0 (zero), the

members of the hunt group ring until the call is answered or the caller disconnects.

•

ext-fwd is the number to which the call is forwarded after the expiration specified by max-hunt.

One series of parameters define a single hunt group. You can define multiple hunt groups separated by

a vertical bar (|). the commas, colons, and semi-colons shown in the syntax and examples are required.

For example, the following script configures two hunt groups:

6001:name=TechSupport,5021,5020,5019,hunt=ne;6;3,cfwd=aa|6002:name=Sales,5013,5014,5026,hu

nt=ra;4;1,cfwd=vm25026

The first hunt group, assigned to extension 6001, is called TechSupport. It contains two extensions, 5021

and 5019, which ring sequentially. The interval before the second extension rings is 6 seconds, and the

entire hunt group is tried three times before it is forwarded to the Auto-Attendant.

The second hunt group, assigned to extension 6002, is called Sales. It contains three extensions, 5013,

5014, and 5026, which ring in random order. Each extension rings for four seconds before the next

extension is tried, and the entire cycle is only repeated once before the call is forwarded to the voicemail

server vm25026.

Using the Contact List Parameter

You can also use the <Contact List> parameter to specify a hunting rule. Note that <Contact List> is a

per line interface parameter and is applicable to that line interface only.

Where:

•

rule = [did:]ext[,ext[,ext[ ]]][,name=gname][,hunt=hrule][,cfwd=target]

did = Embedded DID number. If did is not specified, the rule applies to any DID number

ext = Client extension number pattern; may contain * or ? wildcard or %xx escaped characters

target = a user-id to forward the caller if no one replies or the client stations has rung for a delay

equal to the value specified in <Cfwd No Ans Delay>.

If target starts with “vm”, such as “vm3456”, the call is forwarded to voicemail with the mailbox ID

equal to the rest of the digits. If “,cfwd=…” is not present, the call fails without being forwarded

and <Cfwd No Ans Delay> does not apply.

•

name = A name to associate with the hunt group.

Each member of a rule can be the extension of a group defined in <Hunt Group>. For example,

530?,hunt=ra;10;1,cfwd=vm5404

5300, cfwd=vm5300

4089993001:5001|4089993000:5000,cfwd=aa

Using the Administration Web Server to Configure Hunt Groups

Note

To configure a hunt group, you can use the SPA9000 Setup Wizard, but if you changed the SPA9000

configuration using the administration web server, these changes might be lost if you subsequently use

the Setup Wizard.

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Chapter 4 Configuring SPA9000 Features

Managing Call Forwarding

To use the administration web server to configure the hunt group, complete the following steps:

Step 1

Direct your browser to the following URL:

spa9000_ip/admin/voice/advanced

Replace spa9000_ip with the IP address of the SPA9000. Use IVR option 110# to determine the IP

address. Before using the administration web server, you must enable it on the WAN interface, using

option 7932#.

You can also negotiate to this page from the first page of the administration web server by clicking

Admin Login, Advanced, and then click the Voice tab.

The system prompts for the Administrator account password if it has been set. After authentication, if

required, the Voice - Info page appears.

Step 2

Click the SIP tab and scroll down to the PBX Parameters section.

This section is illustrated in Figure 4-14.

Figure 4-13 PBX Param eters

Step 3

Enter the hunt group script into the Hunt Groups field.

The syntax for the hunt group script is as follows:

extension-in:name=hunt-groupname,hunt-list,hunt=style;ring-interval;max_hunt,cfwd=extensio

n-fwd

For details and an example hunt group rule, see the “Configuring a Hunt Group Rule” section on

page 4-19.

Step 4

Click Submit All Changes.

Client Station Blind Transfers External Caller To DID/Hunt Group

This is a special case where a client station can transfer an external caller on a line interface to a hunt

group defined in the line <Contact List>. Typically, the client station is an AA that prompts the external

caller to enter the desired DID (hunt) group number.

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For example, on Line 1, you can define one hunt group with the virtual extension 50300 for Sales, and

another with extension 50400 for Customer Support. In this example, Line 1 has the main number of

14083331000. The <Contact List> in this example looks like the following:

14083331000:aa|50300:5001,5002,hunt=next;10;1|50400:5003,5004,5005,hunt=random;15;0

In this example, when an external caller calls 14083331000, the AA answers and prompts the caller to

enter a target extension. If the caller enters 50300, the AA blind transfers to the 50300 hunt group, which

consists of two stations. If the caller enters 50400, the AA blind transfers to the 50400 hunt group, which

consists of three stations.

Note

In this example, only 14083331000 is assigned by the ITSP. The two hunt group extensions are virtual

and can be selected by the SPA9000 administrator.

Using Shared Line Appearance

Shared line appearance (SLA) allows multiple stations to share an extension number and manage a call

as a group. One of the most interesting features of SLA is that when the active station places the call on

a shared line key on hold, the call can be resumed from any of the sharing client stations simply by

pressing the corresponding line key from another station.

To configure a group of client stations to share the same extension, perform the following:

•

Select one of the extensions on the client station to configure the share line. As a convention,

Linksys recommends always reserving EXT 1 on the client station as the primary and private

extension of the designated user. Any shared extension should be configured on EXT 2–4.

On the selected EXT, set <Shared Ext> to yes and the <Shared User ID> to the user-id being shared.

Set up the rest of the account information just like the primary EXT on the same station (<User ID>,

<Password>, <Proxy>, and so on).

Assign one or more line key buttons on the client station to the shared extension. Set <Shared Line>

of the line key to “yes”.

It may be more convenient to think of a shared line appearance as a physical line key on the client station.

Sharing line appearance is just like sharing line keys across multiple client stations: if a shared line key

is being used by one of the stations in the group, the other members in the group cannot use the line key

until it is released. All the stations sharing the line key, however, can monitor the status of the line key,

which includes such information as the following:

•

Is the line key free?

Is the line key ringing?

Who is using the line key, and to whom is the user talking?

When there is an incoming call to the shared extension, the SPA9000 alerts all the sharing client stations.

If a client station happened to have both share and private line keys assigned to the shared extension, a

shared line key is chosen first to receive the call before any private line keys.

SPA9000 acts as the state agent (sa) to support SLA. Client stations send NOTIFY (with the dialog event

package and dialog-info+xml message body) to SPA9000 to inform any changes to a shared line key.

The SPA9000 relays the state to all the rest of the sharing client stations with a similar NOTIFY. All the

NOTIFY messages are unsolicited to eliminate the burden of maintaining subscription dialogs. The state

of the dialog-info XML is set to “partial” to indicate this is the state change pertaining to one line key

only.

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Managing Call Forwarding

When a client station boots up and successfully registers with the proxy of each shared EXT, it sends a

NOTIFY of dialog-info to the SPA9000 for the first time. The state of this dialog-info is set to “full.” In

this case, all the dialog states should be idle because the client station has just booted. This indicates the

station is just starting up. In response, the SPA9000 sends a full-state dialog-info NOTIFY to the client

station with the current states of the dialog on each of the share line keys for this share extension.

The example below shows extension 5041 shared by the User-A station (whose primary extension is

5031), with no one using any of the share line appearances on the shared extension.

1. Client station initial NOTIFY/dialog to sa:

NOTIFY sip:[email protected]:6060 SIP/2.0Via: SIP/2.0/UDP

192.168.0.4:5061;branch=z9hG4bK-aba99b57From: “User-A”

<sip:[email protected]:6060>;tag=e676b37bae382029o1To: “User-A”

<sip:[email protected]:6060>Call-ID: [email protected]: 64949 NOTIFY

Max-Forwards: 70

Contact: “User-A” <sip:[email protected]:5061>Event: dialog

Content-Length: 155

Content-Type: application/dialog-info+xml

<?xml version=”1.0”?>

<dialog-info xmlns=”urn:ietf:params:xml:ns:dialog-info” version=”0” state=”full”

entity=”sip:[email protected]:6060”>

</dialog-info>

2. SPA9000 initial NOTIFY/dialog to client station:

NOTIFY sip:[email protected]:5061 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK-512b4023

From: PBX Line 1 <sip:[email protected]>;tag=5080117180da6835o2

To: <sip:[email protected]:5061>

Call-ID: ebec254f-a8610a68@localhost

CSeq: 54508 NOTIFY

Max-Forwards: 70Event: dialogContent-Length: 151

Content-Type: application/dialog-info+xml

<?xml version=”1.0”?>

<dialog-info xmlns=”urn:ietf:params:xml:ns:dialog-info” version=”30” state=”full”

entity=”[email protected]:6060”></dialog-info>

When a client station sends NOTIFY to the SPA9000, the user-id field of the Request-URI must be “sa”

and the message must include a CONTACT header. The CONTACT must be the same as the one used

for registration. The SPA9000 tracks the client station that is using a share line key based on the

CONTACT information in the NOTIFY request sent by client stations.

Before a client station can seize a share line key to make a call, it must send a partial-state dialog-info

NOTIFY with the dialog state of the corresponding line key set to “trying.” The SPA9000 either replies

202 to the NOTIFY if no one is using that line key, or 403 if the line key is currently used by another

client station.

In the example below, User-A is trying to use the first shared line key (x-line-id = 0) to make a call.

Because no one is using that line key, the sa replies 202 to the NOTIFY. Subsequently, SPA9000 notifies

the other client stations sharing that line key so that they would not attempt to use that line key (but can

monitor its status).

1. User-A tries to make a call on the first shared line key on the shared extension 5041:

NOTIFY sip:[email protected]:6060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK-55f99f8e

From: “User-A” <sip:[email protected]:6060>;tag=e676b37bae382029o1

To: “User-A” <sip:[email protected]:6060>

Call-ID: [email protected]

CSeq: 6583 NOTIFY

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Chapter 4 Configuring SPA9000 Features

Managing Call Forwarding

Max-Forwards: 70

Contact: “User-A” <sip:[email protected]:5061>Event: dialog Content-Length: 408

Content-Type: application/dialog-info+xml

<?xml version=”1.0”?>

<dialog-info xmlns=”urn:ietf:params:xml:ns:dialog-info” version=”0” state=”partial”

entity=”sip:[email protected]:6060”>

<local>

<identity display=”User-A”>sip:[email protected]:6060</identity>

</target>

</local>

<state>trying</state>

</dialog>

</dialog-info>

2. The SPA9000 grants the line seize request by replying 202 (because no one else is using that line

key):

SIP/2.0 202 Accepted

To: “User-A” <sip:[email protected]:6060>;tag=41a7-0

From: “User-A” <sip:[email protected]:6060>;tag=e676b37bae382029o1

Call-ID: [email protected]

CSeq: 6583 NOTIFY

Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK-55f99f8e

Server: Sipura/SPA2100-3.1.5(a)

Content-Length: 0

3. The SPA9000 notifies other client stations sharing the line key on the shared extension:

NOTIFY sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK-1cae6e99

From: PBX Line 1 <sip:[email protected]>;tag=5080117180da6835o2

To: <sip:[email protected]:5060>

Call-ID: 3ecb02c9-250f41cd@localhost

CSeq: 56195 NOTIFYMax-Forwards: 70Event: dialog

Content-Length: 397

Content-Type: application/dialog-info+xml

<?xml version=”1.0”?>

<dialog-info xmlns=”urn:ietf:params:xml:ns:dialog-info” version=”31” state=”partial”

entity=”[email protected]:6060”> <dialog id=”201” direction=”initiator”>

<local>

<identity display=”User-A”>sip:[email protected]:6060</identity>

</target>

</local>

<state>trying</state>

</dialog>

</dialog-info>

When a client station sends the initial NOTIFY to the SPA9000, all the line key resources allocated to

this station, if any, are released immediately to avoid locking up of line keys because of client reboot.

The same is done when the registration of the client station expires.

Note that the SPA9000 assumes all partial-state dialog-info and contains only one dialog.

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Chapter 4 Configuring SPA9000 Features

Managing Call Pickup

This section describes the features available for picking up calls and includes the following topics;

•

Directed Call Pickup, page 4-25

Group Call Pickup, page 4-26

Call Park and Pickup, page 4-26

Directed Call Pickup and Group Call Pickup let the user answer an incoming call to a client station from

another client station.

Directed Call Pickup

To use Directed Call Pickup, the user performs the following steps.

1. Determine which client station is ringing (by listening, for example). This is the target station.

2. Press the “pickup” softkey or dial the corresponding *code on the client station.

3. Wait for the client station prompt to enter the target station number.

4. Enter either the extension number of the target station or the <Station Name> of the station.

5. The client station then displays a menu of calls that are currently ringing on the target station.

6. Select the call to pickup from the menu.

The result is that the user is connected to the caller of the selected incoming call. The target station stops

ringing (unless it has more incoming calls).

It is usually more convenient to dial the extension number of the target instead of its <Station Name>.

The main drawback of this approach is that with multiple extension numbers on the target station the

user does not know which extension is ringing. Dialing the <Station Name> retrieves all the calls ringing

on all the extension numbers on the target station.

One workaround is to assign a simple numeric value to <Station Name> for each client station (which

must not conflict with the actual extension numbers used in the PBX). A better alternative is to invoke

the “Corporate Directory” application of the SPA9000 when prompted to enter the target station number

by pressing the “dir” softkey and selecting the “Corporate Directory” option.

The corporate directory includes the <Station Name> of all the client stations that are currently

registered with the SPA9000. This method still assumes that the user knows the <Station Name> of the

target station. The administrator should assign a <Station Name> to each client station that can easily be

remembered, such as the first name of the primary user of that station.

Note

Spaces are not allowed in the <Station Name> parameter. By default, <Station Name> is set to the MAC

address of the client station when auto-provisioned by the SPA9000.

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Chapter 4 Configuring SPA9000 Features

Managing Call Pickup

Group Call Pickup

To use Group Call Pickup, the user performs the following steps.

1. Determine whether any client station is ringing (by listening, for example). It could be more than

one.

2. Press the “GrPick” softkey or dial the corresponding *code on the client station.

3. Wait for the client station to return a list of <Station Name> of all the client stations that are

currently ringing. Note that the SPA9000 does not include the client station in this list.

4. Select the target station that you want to pickup.

5. Wait for the client station to display a menu of calls that are currently ringing on the target station.

6. Select the call to pickup from the menu.

The result is that the user is connected with the caller. The target station stops ringing (unless it has more

incoming calls).

Call Park and Pickup

Call park/pickup allows the user to park a call on one client station and pick it up from the same or a

different client station.

To park a call, the user performs the following steps.

1. While the call is connected, press the “park” softkey on the client station or dial the corresponding

IVR code.

2. Enter the parking lot number to park the call.

The result is that the call is parked and the user is disconnected from the call.

Note the following:

•

Parking a call is very similar to a blind transfer operation; you basically blind transfer the caller to

the parking lot.

If the parking lot number is already in use, the user is notified with an error message on the display.

The user may choose to re-enter a different parking lot number at the point.

Parking lot number cannot be 0.

To pick up a parked call, the user performs the following steps.

1. Press the “unpark” softkey or dial the corresponding *code on the client station.

2. Enter the parking lot number where the call is parked.

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Chapter 4 Configuring SPA9000 Features

Multicast and Group Paging

The result is the user is connected with the parked call. If the parking lot number is not valid, the

operation fails.

The parking lot is implemented as an internal SIP client station of the SPA9000. It has the special user-id

callpark and can handle up to 10 calls. You can specify a music on hold (MOH) server to be used with

call parking by specifying it in the <Call Park MOH Server> parameter (which by default is “imusic”,

the internal music source). If <Call Park MOH Server> is not specified or invalid, the parked call hears

silence while the call is parked.

Multicast and Group Paging

The SPA9000 communicates with all the client stations at once using IP multicast. The multicast address

is specified using the <Multicast Address> parameter for both the SPA9000 and client stations. The

default value is 224.168.168.168:6061.

Note

Make sure that the SPA9000 and the SPA900 Series phones use the same multicast address and port

number. For the SPA9000 and the SPA901 phone, use the IVR to identify or change the current address

(180# and 181# respectively). For other SPA900 Series phones, press Menu and select Network > 15

Multicast Address.

The SPA9000 can send the following messages to the group:

•

Graceful reboot

Immediate reboot

Graceful restart

Immediate restart

Group page start

Group page end

Get ringing calls

Client stations send the following message to the SPA9000:

Looking for configuration server

•

When the SPA9000 initially boots up, it sends a multicast message to request all client stations to

gracefully reboot. This allows the client stations to redo DHCP discovery, download the latest phone

parameters, and also register with the SPA9000. The SPA9000 does not store any DHCP lease states and

client registration states in Flash memory.

In addition to group management, IP multicast is also used in the group paging application, where the

announcer sends RTP packets to an IP multicast address at which all the other client stations are

listening. This address is chosen by the SPA9000 and is configured using the <Group> parameter. The

default value is 224.168.168.168:34567.

Group paging lets the user page all the client stations at once. If the client station is on a call while a

group page starts, the call is automatically placed on hold. The speaker on the paged stations is turned

on automatically unless the handset or headset is being used. Group page is one-way only. The paged

client stations can only listen to the call from the originator.

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Chapter 4 Configuring SPA9000 Features

Music On Hold

To start a group page, make a call by dialing pagegroup. For convenience, this special number is

included in the corporate directory. A speed dial or personal directory entry can also be defined locally

on a specific phone.

Music On Hold

This section describes how to manage the music on hold (MOH) feature. It includes the following topics:

•

Overview, page 4-28

Changing the Internal Music Source, page 4-28

Restoring the Default Internal Music Source, page 4-29

Using a Streaming Audio Server, page 4-30

Using the IVR with an SAS Line, page 4-30

Example SAS with MOH, page 4-31

Configuring the Streaming Audio Server, page 4-32

Overview

The SPA9000 has an internal music source with the user-id imusic that plays an internally stored music

file repeatedly. The SPA9000 ships with a default music file (Romance de Amor). You can override this

file by downloading a new file into the unit using TFTP. You do this by entering a valid value in the

<Internal Music URL> parameter. The music file must be in G.711u format sampled at 8000 samples/sec

with no file header. The maximum length is 65.5 seconds.

You can control the way the music file is played by setting the <Internal Music Script> parameter. To

use the internal music source as the MOH server on a client station, specify imusic for the value of the

<MOH Server> parameter on the client station.

Changing the Internal Music Source

The following resources are required for changing the internal music source:

TFTP server software

The IP address of the computer on the LAN connected to the SPA9000.

Music source in G.711 u-law format at 8000 samples per second with no header information.

•

Before you begin, make sure the TFTP server software is running on your computer. Then complete the

following steps.

Step 1

Step 2

Direct your browser to the following URL:

spa9000_ip/admin/voice/advanced

Replace spa9000_ip with the IP address of the SPA9000. Use IVR option 110# to determine the IP

address.

Click the SIP tab and scroll down to the Internal Music Source Parameters section.

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Chapter 4 Configuring SPA9000 Features

Music On Hold

This section is illustrated in Figure 4-14.

Figure 4-14 Internal Music Source Param eters

Step 3

Enter tftp://server_IP_address:port/path in the Internal Music URL field.

Replace server_IP_address with the local IP address of the computer you are using as the TFTP server.

Replace port with the port number used by the TFTP server; by default it is 69. Replace path with the

location and name of a music file in the correct format.

For example, if the computer local IP address is 192.168.0.5, the directory is named musicdir, and the

converted music file is named jazzmusic.dat, then tftp://192.168.0.5:69/musicdir/jazzmusic.dat should

be entered in the Internal Music URL field. (The default port number, 69, is used.)

Step 4

Click Submit All Changes.

The SPA9000 reboots.

After rebooting, the SPA9000 downloads the file and stores it in flash memory.

Restoring the Default Internal Music Source

To restore the original default music file for the internal music source, perform the following steps:

Step 1

Download the file from the following website:

http://www.linksys.com/download/spa9000/romance711u2.zip

Unzip this file and save it to an accessible directory on your TFTP server.

Direct your browser to the following URL:

Step 2

Step 3

spa9000_ip/admin/voice/advanced

Step 4

Step 5

Click the SIP tab and scroll down to the Internal Music Source Parameters section.

Enter tftp://server_IP_address:port/path/romance711u2.zip in the Internal Music URL field.

The server_IP_address is the local IP address of the computer you are using as the TFTP server. The

port is the port number used by the TFTP server; by default it is 69. The path indicates the location and

name of the music file.

Step 6

Click Submit All Changes.

The SPA9000 reboots.

After rebooting, the SPA9000 downloads the file and stores it in flash memory.

This section describes how to use and configure a streaming audio server (SAS). It includes the

following topics:

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Chapter 4 Configuring SPA9000 Features

Music On Hold

•

Using a Streaming Audio Server, page 4-30

Using the IVR with an SAS Line, page 4-30

Example SAS with MOH, page 4-31

Configuring the Streaming Audio Server, page 4-32

Using a Streaming Audio Server

On a connected call, the SPA9000 may place the remote party on hold by performing a hook-flash to

initiate a three-way call or by swapping two calls during call-waiting. If the remote client indicates that

it can still receive audio while the call is holding, the SPA9000 can be configured to contact an

auto-answering streaming audio server (SAS) to stream audio to the holding party. When used this way,

the SAS is referred to as an MOH Server.

The SAS feature lets you use attach an audio source to one of the SPA9000 FXS ports (Phone 1 or Phone

2 on the PAP2T) and use it as a streaming audio source device. If the SPA9000 has multiple FXS ports,

either or both of the associated lines (Line 1 and Line 2 on the PAP2T) can be configured as an SAS

server.

When the line is called and the FXS port is off hook, the SPA9000 answers the call automatically and

streams audio to the calling party.

If the FXS port is on-hook when the incoming call arrives, the SPA9000 replies with a SIP 503 response

code to indicate “Service Not Available.” If an incoming call is auto-answered, but later the FXS port

changes to on-hook, the SPA does not terminate the call but continues to stream silence packets to the

caller. If an incoming call arrives when the SAS line has reached full capacity, the SPA replies with a

SIP 486 response code to indicate “Busy Here.”

The SAS line can be setup to refresh each streaming audio session periodically using a SIP re-INVITE

message, which detects if the connection to the caller is down. If the caller does not respond to the

refresh message, the SAS line terminates the call so that the streaming resource can be used for other

callers.

Each SAS server can maintain up to five simultaneous calls. If the second line on the SPA9000 is

disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls will receive

a busy signal (SIP 486 Response).

The streaming audio source must be off-hook for the streaming to occur. Otherwise incoming calls will

get a error response (SIP 503 Response). The SAS line will not ring for incoming calls even if the

attached equipment is on-hook.

If no calls are in session, battery is removed from tip-and-ring of the FXS port. Some audio source

devices have an LED to indicate the battery status. This can be used as a visual indication as to whether

audio streaming is in progress.

Set up the Proxy and Subscriber Information for the SAS Line as you normally would with a regular user

account.

Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery features are not available

on an SAS line.

Using the IVR with an SAS Line

The IVR can still be used on an SAS line, but the user needs to follow the following steps:

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Chapter 4 Configuring SPA9000 Features

Music On Hold

Step 1

Step 2

Step 3

Step 4

Power off the SPA9000.

Connect a phone to the port and make sure the phone is on-hook.

Power on the SPA9000.

Pick up handset and press * * * * to invoke IVR in the usual way.

If the SPA9000 boots and finds that the SAS line is on-hook, it will not remove battery from the line so

that IVR may be used. But if the SPA9000 boots up and finds that the SAS line is off-hook, it will remove

battery from the line because no audio session is in progress.

Example SAS with MOH

Figure 4-15 MOH Application w ith a SPA9000 Line Configured as an SAS

SPA2:

IP=192.168.2.200

UserID[1]=2001, SIP Port[1]=5060

UserID[1]=2002, SIP Port[1]=5061

SPA1:

IP=192.168.2.100

UserID[1]=1001, SIP Port[1]=5060

UserID[1]=1002, SIP Port[1]=5061

IP Network

Phone 1

Phone 2

Media

signal

adapter

Line in

Music

source

In this example, the SAS Line is registered with the SPA9000 as the other subscribers.

On SPA1:

SAS Enable[1] = no

MOH Server [1] = 1002

SAS Enable[2] = yes

On SPA2:

SAS Enable[1] = no

MOH Server [1] = 1002

SAS Enable[2] = no

MOH Server [2] = 1002

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Chapter 4 Configuring SPA9000 Features

Enhancements in Release 5.1

Configuring the Streaming Audio Server

The following provides step-by-step procedures for implementing a SAS with an external music source.

Step 1

Step 2

Connect an RJ-11 adapter between the music source and an FXS port on the SPA9000 (Phone 1 or Phone

2).

On the administration web server, click the SIP tab and scroll down to the Streaming Audio Server

section (see Figure 4-16).

Figure 4-16 Enabling the Stream ing Audio Server

Step 3

Step 4

On the SAS Enable pull-down selection list, select yes.

In the MOH Server field of the Call Feature Settings section, enter the User ID configured for the line

attached to the audio source.

Step 5

Click Submit All Changes.

Enhancements in Release 5.1

This section describes feature enhancements introduced in Release 5.1 of the SPA9000. It includes the

following topics:

•

Bridge Mode, page 4-32

Renew DHCP On SIP Request Timeout, page 4-34

In addition, Release 5.1 introduces downloadable prompts for the Auto-Attendant, a feature that is useful

for localization of the SPA9000. For more information about this feature, see the “Downloading

Prompts” section on page 5-8.

Bridge Mode

In Release 5.1, the <CWFD Bridge Mode> and the <XFER Bridge Mode> parameters were added to

improve the way that the SPA9000 handles call forwarding and call transfer. This section describes the

added functionality provided and includes the following topics:

•

Call Forward Bridge Mode, page 4-33

Call Transfer Bridge Mode, page 4-33

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Chapter 4 Configuring SPA9000 Features

Enhancements in Release 5.1

Call Forward Bridge Mode

The normal way of performing the call forwarding operation is for the SPA9000 to send a (blind) SIP

REFER to the calling device to let it contact the target number directly. It then drops out of the call

completely. This requires the calling device to understand the SIP signaling involved and the operation

permitted by the underlying service provider. The SPA400 cannot handle this operation.

With bridging, the SPA9000 maintains two separate call legs throughout the call: one with the caller and

one with the call forward target. The two call peers connect only with the SPA9000, while the SPA9000

acts as a proxy for the RTP packets exchanged between the two parties.

The <CWFD Bridge Mode> parameter has two possible values:

•

none—Do not bridge forwarded calls (use the normal REFER method)

all—Bridge all forwarded calls

Call Transfer Bridge Mode

The normal way of performing this operation is for the SPA9000 to send a SIP REFER method to the

calling device to let it contact the transfer target directly. The SPA9000 then drops out of the call

completely. This requires the calling device (the transferee) and the target device to understand the SIP

signaling involved and the operation permitted by the underlying service providers. Note that the call

legs with transferee and the transfer target might be with different ITSP. The SPA400, for instance,

cannot handle this operation.

With bridging, the SPA9000 maintains two separate call legs throughout the call: one with the

transferred call and one with the transfer target. The two call peers connect only with the SPA9000, while

the SPA9000 acts as a proxy for the RTP packets exchanged between the two parties.

The <XFER Bridge Mode> parameter has three possible values:

•

none —Do not bridge call transfer (use the normal REFER method)

all —Bridge all call transfer

all except same line—Bridge call transfer only if it is between two different line interfaces

This enhancement is related to how the SPA retries the SIP REGISTER operation when the current

In previous versions of the SPA9000, three parameters, available on the SIP tab of the administration

web server, have been used to control the interval between retries after a REGISTER failure:

•

<Reg Retry RSC> = A comma separated list of failure register response codes. When one of these

codes occurs, the SPA9000 waits for a duration equal to the number of seconds defined in the <Reg

Retry Invl> parameter.

For other failure codes, the SPA9000 uses the value specified in the <Reg Retry Long Intvl>

parameter. Note that if the REGISTER request times out waiting for a response, or if the request

encounters an ICMP error, the SPA9000 always uses the value specified in the <Reg Retry Intvl>

parameter

•

<Reg Retry Intvl> = Retry interval in seconds

<Reg Retry Long Intvl> = Long retry interval in seconds

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Chapter 4 Configuring SPA9000 Features

Enhancements in Release 5.1

The first part of the REGISTER enhancement adds a randomized delay on top of the corresponding

nominal retry interval:

•

<Reg Retry Random Delay> = range of randomized delay to add to <Reg Retry Intvl> before

retrying REGISTER (when <Reg Retry Intvl> is applied)

•

<Reg Retry Long Random Delay> = range of randomized delay to add to <Reg Retry Long Intvl)

before retrying REGISTER (when <Reg Retry Long Intvl> is applied)

The second enhancement is to allow exponential back off of the <Reg Retry Intvl> for each retry until

it reaches the maximum value defined in the <Reg Retry Intvl Cap> parameter. If the value of the

<Reg Retry Intvl Cap> parameter is less than the <Reg Retry Intvl), the exponential back off feature is

disabled.

Renew DHCP On SIP Request Timeout

The SPA9000 may sometimes have an invalid IP address even though the DHCP lease has not expired.

In these cases, the SPA9000 may not receive any packets until it renews the IP address lease with the

DHCP server. The Renew DHCP On SIP Request Timeout feature, introduced in Release 5.1, allows the

SPA9000 to discover that the currently assigned IP address is no longer valid. This feature triggers a

DHCP lease renewal whenever indicators suggest that the SPA9000 is not receiving packets.

The SPA9000 periodically sends SIP requests to register, make calls, and so forth. Each SIP request

normally receives a corresponding reply. If the SPA9000 fails to receive a reply to a SIP request this may

indicate an invalid local IP address.

With this enhancement, when the SPA9000 fails to receive a reply to a SIP request, a timer is started. If

no reply to any SIP request is received before the timer expires, the SPA9000 automatically performs a

DHCP lease renewal. If the renewal results in the same IP address, no further action is taken. If the

renewal results in a new IP address assignment, the SPA9000 reboots.

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C H A P T E R

Configuring the LVS Auto-Attendant

This chapter describes how to configure the LVS Auto-Attendant using the IVR, the Setup Wizard, and

XML scripting. This chapter contains the following sections:

•

Configuring Auto-Attendant, page 5-1

Downloading Prompts, page 5-8

Configuring Dial Plans for the Auto-Attendant, page 5-9

Alternative AA Configuration, page 5-10

Switching Between Alternative AAs Using the IVR, page 5-10

XML Scripting for the Auto-Attendant, page 5-13

AA XML Script Examples, page 5-17

Auto-Attendant XML Instructions Set, page 5-22

Configuring Auto-Attendant

This section describes how to configure the prompts and schedule for the Auto-Attendant and includes

the following topics:

•

How the Auto-Attendant Works, page 5-1

Using the IVR to Record Auto-Attendant Prompts, page 5-2

Using the Wizard to Configure the Auto-Attendant, page 5-4

Using the Administration Web Server to Configure the Auto-Attendant, page 5-6

How the Auto-Attendant Works

The Auto-Attendant (aa) is an internal service within the SPA9000. It plays pre-recorded voice messages

that offer the caller a menu of choices, so the Auto-Attendant can appropriately direct the call. For

example, a greeting could be, “Welcome to the abc company. For sales, press 1. For service, press 2. To

speak to our operator, press 3.” (This is a custom greeting, which you could record using the IVR Menu.)

After the caller has made a choice, the call is routed to the appropriate extension.

There are three Auto-Attendants available, one for daytime, one for nighttime, and one for weekend or

holidays.

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Auto-Attendant

The daytime Auto-Attendant is preconfigured and enabled by default, and the first message it plays

(Prompt ID 1) is suitable for business hours.

If you want a caller to hear a different greeting during nighttime (non-business) hours, then you should

record a new prompt, such as Prompt ID 5, using the Interactive Voice Response Menu and then

configure the Auto-Attendant settings using the administration web server. For example, Prompt ID 5

could say, “The company is currently closed. Our business hours are 9 AM to 5 PM, Monday to Friday.”

When the Auto-Attendant is enabled, it parses and operates on user input (key presses or DTMF tones)

following the rules specified in the Auto-Attendant dial plan of the SPA9000. These rules are specified

by the AA Dial Plan parameters found on the Voice - SIP screen of the administration web server. For

information about configuring the AA Dial Plan, refer to the “Configuring Dial Plans for the

Auto-Attendant” section on page 5-9. The Auto-Attendant allows users to save up to ten prompts. By

default, AA fills the first four prompts as listed in Table 5-1.

Table 5-1 Prom pts

Prompts ID

Prompt1

Prompt2

Prompt3

Prompt4

Default Audio Content

“If you know your party’s extension, you may enter it now.”

“Your call has been forwarded.”

“Not a valid extension, please try again.”

“Goodbye”

Users can listen, record, their customized greeting, menu, warning, ending, and so on from IVR, using

code 72255. Users can also erase a recorded prompt from IVR.

The recorded prompt is encoded with G.711U and saved in flash. However, these customized prompts

are erased when a factory reset is performed.

The maximum length of each prompt is one minute. Users can customize up to 94.5 seconds of audio in

addition to the default prompts. When there is not enough memory left, the IVR menu ends the recording

automatically. The user can check the AA prompts status from the Auto-Attendant Prompt Status section

on the administration web server Voice-Info page.

Using the IVR to Record Auto-Attendant Prompts

This section describes how to use the IVR to record the Auto-Attendant prompts using the IVR.

To record Auto-Attendant prompts, complete the following steps from the IVR Menu.

Step 1

Step 2

Using an analog telephones connected to the SPA9000, press **** (in other words, press the star key

four times).

Wait until you hear the following prompt:

“Linksys configuration menu. Please enter the option followed by the # (pound) key or hang up to exit.”

Press 72255# to access the Auto-Attendant message settings.

Step 3

Step 4

Step 5

You hear, “Please enter the message number followed by the # (pound) key.”

Enter the number of the message you wish to record, review, or delete.

You hear, “Enter 1 to record. Enter 2 to review. Enter 3 to delete. Enter * to exit.”

Press 1 on the phone to record a new message.

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Auto-Attendant

When the user presses this option, the IVR checks the available buffer size. If there is no more buffer

capability, IVR plays the “option failed” message and returns to the previous menu.

If the buffer space is sufficient, you hear, “You may record your message after the tone. When

finished, press #.”

Step 6

Step 7

Record the new message and press #.

After you record the message, you hear, “To save, enter 1. To review, enter 2. To re-record, enter 3. To

exit, enter *.”

Press 1 to save the new recorded message.

When recording finishes and the user agrees to save the prompt, if the recorded prompt is longer than 15

seconds, IVR plays a short “one moment, please” message to inform the user to wait for several seconds

while the save is completed.

Figure 5-1 shows the IVR prompt menu call flow.

Figure 5-1 IVR Prom pt Menu Call Flow

User enters

72255 from IVR

Menu-1

Play prompt (“Please enter message

number followed by the # key”)

User enters

prompt key

(1-10)

Menu-2

Goto menu-2

Goto menu-1

User enters *

Play prompt (“Enter 1 to record,

enter 2 to review, enter 3 to

delete, enter * to exit”)

User enters 2

User enters 1

User enters 3

Menu-3-A

Menu-3-B

Retrieve the

prompt data and

play prompt

Play prompt (“You may record your

message after the tone. When

finished, press #.”)

^{Play prompt (“}enter 1 to

^{confirm, enter}* to exit

User enters 1

User enters *

User records

a prompt

Erase message

Menu-4-A

Play prompt (“to save enter 1, to

review enter 2, to rerecord enter

3, to exit enter *.”)

Play prompt

(“Message erased.”)

User enters *

Goto

4-A

User enters 1

User enters 2

User enters 3

Play prompt

(“Value saved”)

Play user

recorded

prompt

Goto menu-3-A

Goto menu-2

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Auto-Attendant

Using the Wizard to Configure the Auto-Attendant

You can use the Setup Wizard to customize a schedule for Auto-Attendant. To do this, complete the

following steps.

Step 1

Select Auto Attendant from the Advanced Feature Menu (see Figure 5-2).

Figure 5-2 Auto Attendant Option

The system displays Page 1, shown in Figure 5-3

Select Auto Attendant from the Advanced Feature Menu (see Figure 5-2).

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Auto-Attendant

Figure 5-3 Auto Attendant Wizard—Page 1

Select Advanced Auto-Attendant and click Next.

The system displays the screen shown in Figure 5-4.

Figure 5-4 Auto Attendant Wizard—Page 2

Step 2

Select the office working and non-working hours from the pull-down selection lists and click Next.

The system displays the screen shown in Figure 5-5.

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Auto-Attendant

Figure 5-5 Auto Attendant Wizard—Page 3

Step 3

Step 4

Select the extensions to ring for the receptionist.

Select the hunt group to ring in response to user input.

Use the Hunt Groups option on the Advanced Features menu to configure the hunt group. For more

information about hunt groups, see the “Configuring a Call Hunt Group” section on page 4-6 and

the“Using Call Hunt Groups” section on page 4-19.

Using the Administration Web Server to Configure the Auto-Attendant

Each message is internally referred to as Prompt x, x being a number ranging from 1 to 10. The following

parameters are the minimum that must be configured.

•

Contact List. This is a list of clients that the SPA9000 alerts when there is an incoming call on the

line. The Auto-Attendant must be included on this list. By default, the Auto-Attendant is the only

client on this list, so the Auto-Attendant picks up every call. You can decide to have the

Auto-Attendant pick up a call if a number (or group) of clients did not pick up the call first. This

parameter is configured on the Voice - Line x (x is 1-4) screen of the administration web server.

•

AA Script. The SPA9000 lets you program the Auto-Attendant instructions using XML scripts.

These parameters, AA Scripts 1-3, are configured on the Voice - SIP tab on the administration web

server. Only one script is active at a time. Scripting is described in more detail below.

AA Dial Plan. The Auto-Attendant parses the user input according to one of the two parameters, AA

Dial Plan 1 or 2. The AA Script includes a reference to one of these two dial plan parameters through

the dial plan instruction. These parameters are configured on the Voice - SIP tab on the

administration web server.

•

DayTime AA Script. This defines which of the three scripts (AA script 1, 2, or 3) should be used for

daytime hours. This parameter is configured on the Voice - SIP screen of the administration web

server.

Figure 5-6 illustrates the Auto-Attendant parameters on the administration web server.

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Configuring Auto-Attendant

Figure 5-6 AA Web Server Configuration Param eters

(50xx|xxxx.)

AA Dial Plan 1:

(<:500>x|50xx.)

AA Dial Plan 2:

<aa><form id="dir" type="menu"><audio src="prompt1" bargein="T"/><noi

<aa><form id="gb" type="node"><audio src="prompt4" bargein="T"/><exit

AA script 1:

AA script 2:

AA script 3:

yes

start=9:0:0;en

DayTime AA:

Day Time:

DayTime AA Script:

NightTime AA:

DayTime Answer Delay:

NightTime AA Script:

Weekend/Holiday AA:

Weekend/Holiday AA Script:

yes

NightTime Answer Delay:

Weekends/Holidays:

Weekend/Holiday Answer Delay:

w k=6,7;hd=1/

The following instructions explain how to use the administration web server to configure the nighttime

Auto-Attendant to use Prompt ID 5 as the initial greeting. You can also use these instructions to record

additional prompts and further customize your Auto-Attendant, as long as you also make the appropriate

changes to the sample AA script 2 code.

To configure the nighttime Auto-Attendant, complete the following steps:

Step 1

Direct a browser on the administration computer to the following URL:

http://spa_IP address/admin/voice/advanced

Step 2

Step 3

Step 4

The Voice - Info screen appears. Click the SIP tab.

On the SIP screen, scroll down to the Auto Attendant Parameters.

Copy the default AA script 1 text to the Notepad (or other word processing program).

The following is the default AA script 1 code:

<aa>

</nomatch>

<match>

</default>

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Chapter 5 Configuring the LVS Auto-Attendant

Downloading Prompts

</match>

</form>

</aa>

Step 5

In the text editor, replace “dir” with “nt” (see the new text in boldface). Then add this line of code:

This is the resulting AA script 2 code:

<aa>

</nomatch>

<match>

</default>

</match>

</form>

</aa>

Step 6

Step 7

Copy the AA script 2 code from the Notepad, and paste it into the AA script 2 field.

For the DayTime field, enter the daytime hours for the daytime Auto-Attendant in 24-hour format. The

start and end times should be in this format:

start=hh:mm:ss;end=hh:mm:ss

(hh for hours, mm for minutes, and ss for seconds)

For example, start=9:0:0;end=17:0:0 means the start time is 9 AM and the end time is 5 PM. The other

hours (5 PM to 9 AM) are considered nighttime hours.

Step 8

Step 9

For the NightTime AA setting, select yes.

For the NightTime AA Script setting, select 2.

Click the Submit All Changes button to save your new settings.

Downloading Prompts

To erase or download customized prompt files for the Auto-Attendant, use the <AA Prompts URL

Script> parameter on the Auto Attendant Parameters section of the administration web server.

The parameters in this section instruct the SPA9000 to download prompt files from a

TFTP/HTTP/HTTPS server. These files must be encoded in G.711u, size less than 60 seconds, with the

header removed.

The total prompt file size cannot be larger than 94.5 seconds. The prompt is downloaded at the device

boot up time. If the prompt has already been downloaded from the given URL, the download is not

performed. If the prompt file name is none, the corresponding prompt currently saved in the flash is

erased. The default value is blank.

The following is the format of the prompt file:

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Chapter 5 Configuring the LVS Auto-Attendant

Configuring Dial Plans for the Auto-Attendant

serv=scheme://server_addr[:port]/root_path;[p1={prompt1 file path name};][p2={prompt2 file

path name};][p3={prompt3 file path name};][p4={prompt4 file path name};][p5={prompt5 file

path name};][p6={prompt6 file path name};][p7={prompt7 file path name};][p8={prompt8 file

path name};][p9={prompt9 file path name};][p10={prompt10 file path name};]

Where:

•

scheme = tftp|http|https

default port is 69 for tftp, 80 for http, and 443 for https

root_path can be empty

[ ] denotes optional item

Example 1:

serv=tftp://192.168.2.150/root/test/;p1=menu.wav;p2=transfer.wav;p3=nomatch.wav;p4=none;

In this example, prompt 1 is downloaded from tftp://192.168.2.150/root/test/menu.wav, prompt

2 from tftp://192.168.2.150/root/test/transfer.wav, prompt 3 from

tftp://192.168.2.150/root/test/nomatch.wav, and prompt 4 is erased.

Example 2:

serv=tftp://192.168.2.150/tt;p1=test/menu.wav;p3=transfer.wav;p5=/nomatch.wav;p7=none;

In this example, prompt 1 is downloaded from tftp://192.168.2.150/tttest/menu.wav, prompt 3

from tftp://192.168.2.150/tttransfer.wav, prompt 5 from

tftp://192.168.2.150/tt/nomatch.wav, and prompt 7 is erased.

Configuring Dial Plans for the Auto-Attendant

The Auto-Attendant (AA) can handle up to ten incoming calls simultaneously. The AA is assigned the

reserved user-ID “aa”. AA is able to automatically answer the outside/inside incoming calls with

greeting messages, directory messages that allow callers to select the appropriate departments or to reach

known extension numbers.

AA is able to accommodate two categories of callers:

•

Callers who are not accustomed to using Auto-Attendants—AA plays greeting and menu messages

to callers and explains how to reach their desired extension.

•

Callers who call you regularly, are accustomed to using Auto-Attendants, know how your system

works, and want to move through it quickly—AA allows experienced users to dial their desired

party while the menu message is still playing.

When the caller inputs DTMF digits, the Auto-Attendant parses them using the dial plan first, and then the

parsing result are directed to the Auto-Attendant script menu instruction. Each AA menu has a dial plan.

Users can define the dial rule in the <AA Dial Plan 1> or <AA Dial Plan 2> parameters. Each AA dial

plan parameter has a matching ID that can be used in the AA XML script. For example, the user can

specify “dp1” to indicate “AA Dial Plan 1”. (See Table 5-2.)

Table 5-2 Param eters and Matching IDs

Web UI Parameter

“AA Dial Plan 1”

“AA Dial Plan 2”

Matching ID in AA Script

“dp1”

“dp2”

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Alternative AA Configuration

The user can define the dial rule in the dial plan parameter, and then do the translation in the AA script.

In this case, the dial plan can be very simple, such as “(1|2|3|4|5xxx)”, “(xxxx|*|#)”. In the AA script,

the user defines how to translate 1|2|3|4 |*|# into the extension number they represent.

The user can also perform the translation in the dial plan first. This dial plan is more complicated, but a

lot of work is saved by using the AA script. Linksys recommends that the user use this method. For

example, the dial plan can be as follows:

(<x:500x>|408555xxxx|xxxxx)”, “(<1:1002>|<2:21111>|<3:3333>|xxxxx)

In this example, when the user inputs DTMF digits, AA parses them using the dial plan first, then the

parsing result is directed to the AA script menu instruction.

Each Auto-Attendant menu has a dial plan. You can define the dial rule by setting the <AA Dial Plan 1> and

<AA Dial Plan 2> parameters. Each Auto-Attendant dial plan setting has a matching ID, which can be used

in Auto-Attendant XML scripting. For example, a user specifies the value dp1 to indicate AA Dial Plan 1.

Alternative AA Configuration

The SPA9000 also supports nighttime AA, weekend AA, and holiday AA treatments. (See Figure 5-6.)

When an alternate AA treatment is enabled, AA checks the current local time with the corresponding

AA date/time range and decides which AA script and answer delay to use.

The user needs to generate additional AA scripts for the weekend AA or holiday AA. The off-office AA

script can be a very simple node form script.

For example:

<aa>

<exit/>

</form>

</aa>

In this example, when a caller reaches the company at off hours, AA picks up the call and plays audio

“prompt4,” and ends the call automatically by executing the exit/instruction.

Switching Between Alternative AAs Using the IVR

The PBX administrator can manually change the current AA treatment using IVR option 79228# (see

Table 5-3). The user needs to turn on the corresponding AA service to make this setting take effect.

•

0—For auto AA treatment based on the <Day Time> <Weekends/Holidays> setting

1—For day time AA treatment

2—For night time AA treatment

3—For weekend/holiday AA treatment

If the corresponding AA service is not enabled, or there is no valid AA script for the specified AA

service, changing the AA treatment has no effect. In this case, the value of the <Current AA> parameter

on the Voice - Info page is Inactive.

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Chapter 5 Configuring the LVS Auto-Attendant

Switching Between Alternative AAs Using the IVR

Table 5-3 Param eter Descriptions

Parameter

Description

AA Dial Plan 1/2

This describes the dial plan rule that the Auto-Attendant uses in a particular

<form>.

AA Script 1/2/3

There are three parameters for storing Auto-Attendant scripts. One option is to

use these as different treatments for daytime hours, nighttime hours, and

weekends/holidays.

Daytime AA

Each parameter controls whether the corresponding Auto-Attendant service is

enabled or disabled. The corresponding IVR code is 79228#

Nighttime AA

Weekend/Holiday AA

Daytime AA Script

Nighttime AA Script

This parameter specifies which script is used for a particular Auto-Attendant

treatment. The choices are 1 for the AA script 1 setting, 2 for the AA script 2

setting, and 3 for the AA script 3 setting.

Weekend/HolidayAA

Script

DayTime Answer

Delay

Each Auto-Attendant service has a corresponding delay setting, which is the

number of seconds you want the Auto-Attendant to wait before answering. By

default, the DayTime Answer Delay setting is 12 seconds. The NightTime and

Weekend/Holiday Answer Delay settings are set to 0 (the Auto-Attendant

answers immediately).

NightTime Answer

Delay

Weekend/Holiday

Delay

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Chapter 5 Configuring the LVS Auto-Attendant

Switching Between Alternative AAs Using the IVR

Table 5-3 Param eter Descriptions

Weekends/Holidays This parameter defines the dates of weekends and holidays.

Format:

[wk=n1[,ni];][hd=mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy[,

mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy];]

The abbreviation wk stands for weekend. The value can be any number, 1 to 7,

to represent Monday to Sunday. Up to four weekend days can be defined.

The abbreviation hd stands for holiday. You do not have to include the year

(yyyy) if you want the same date to apply every year.

Example:

wk=6,7;hd=1/1,7/4

In this example, the weekend includes Saturday (6) and Sunday (7), and the

holidays are January 1st and July 4th every year.

DayTime

This parameter defines the daytime hours. (The other hours are considered

nighttime hours.)

Format:

start=hh:mm:ss;end=hh:mm:ss

The abbreviation hh stands for hours and uses the 24-hour format. The

abbreviation mm stands for minutes, and the abbreviation ss stands for seconds.

Example:

start=9:0:0;end=17:0:0

In this example, the start time is 9 AM and the end time is 5 PM. The other

hours (5 PM to 9 AM) are considered nighttime hours.

If you do not enter start and end times, then the whole day (24 hours) is

considered as daytime, so the nighttime Auto-Attendant is disabled, even if the

NightTime AA parameter is set to yes (enabled).

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Chapter 5 Configuring the LVS Auto-Attendant

XML Scripting for the Auto-Attendant

The SPA9000 AA allows users to define the AA instructions using XML script. This section includes

the following topics:

•

Overview, page 5-13

XML Scripting Grammar, page 5-13

AA Instructions , page 5-15

AA XML Script Examples, page 5-17

Auto-Attendant XML Instructions Set, page 5-22

Example 1—AA Default XML Script, page 5-17

Overview

AA XML scripting has two kinds of dialogs: node type and menu type. The dialog type is specified in

the type attribute of <form> element. The node type dialog is used to execute actions. It cannot recognize

user input. Users must specify an action in node form. The menu type dialog accepts DTMF inputs. AA

recognizes them and performs the corresponding actions.

AA is always in a specific dialog type when it is activated and can be transferred to the other dialog type

with or without user input.

A dialplan statement and a menu instruction must be included in a menu type dialog. The dialplan

statement specifies the dial rule that the menu need to follow. The menu instruction defines what action

the AA needs to perform when user input hits one of the dial rules.

Many parameters can be changed to customize the Auto-Attendant. One of the most important parameters is

the script, or set of instructions, that the Auto-Attendant executes when it is running.

Note

AA is disabled if it encounters an XML script error. If the value of the <Current AA> parameter on the

Info page is Inactive, it may indicate an XML script error. To determine if the AA script is working, test

your script with a SPA900 Series phone. To hear the current AA script, use the Directory softkey and

select Corporate Directory > AA.

XML Scripting Grammar

SPA9000 lets you use XML scripting grammar to define the Auto-Attendant instructions. You have a

choice of three scripts, which are stored in the AA script 1-3 parameters on the Voice - SIP screen of the

administration web server. The instructions must be defined or encapsulated in a <form> structure. You

may have multiple <form> structures within a script that the Auto-Attendant can transfer to, based on

user input.

The XML scripting grammar supports two types of <form> structures, node and menu. The main

difference between the two types is that within the node type, user input cannot be processed—only

actions may be specified.

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Chapter 5 Configuring the LVS Auto-Attendant

XML Scripting for the Auto-Attendant

Note

The maximum compiled size of an AA script is 2048 bytes. Any script longer than this is truncated and

generates an XML script error. Note that when the script is compiled it can grow in length, so this error

can occur even if the script is considerably smaller than 2048 bytes.

Node Type Dialog

The format of the node type dialog is as follows:

</form>

In the node dialog, AA executes the audio instruction first if it is not empty. If there are multiple audio

instructions, AA plays each prompt one by one. When finished playing, AA executes an action

instruction. The action instruction cannot be empty.

Menu Type Dialog

In the menu type, user input can be processed. It is processed according to the dial plan statement

associated with the menu and defines what action the Auto-Attendant executes when the user input

matches the dial plan. The format of the menu type is as follows:

The menu dialog can have five kinds of instructions. The order of the instructions does not affect AA

behavior. Only the order of audio instructions affects the audio playing order.

</form>

Dialplan Statement

The following dialplan statement determines the dialplan ID that the current menu dialog follows.

The user input is passed by the dial plan first, then the passed result is fed into the menu instruction if

the input matches a dial rule. Otherwise, AA executes the nomatch instruction, unless that instruction is

empty.

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Chapter 5 Configuring the LVS Auto-Attendant

XML Scripting for the Auto-Attendant

AA Instructions

This section includes the following topics:

•

Audio Instruction, page 5-15

Action Instruction, page 5-15

Noinput Instruction , page 5-15

Nomatch Instruction, page 5-16

Menu Matched Instruction—Recognition of Touch Tone (DMTP) Key Presses, page 5-16

Audio Instruction

The following is an example of the audio instruction:

AA plays the audio file specified in the src attribute. When playing the audio, AA allows the caller to

interrupt the current prompt by pressing digits when the bargein attribute is set to T. AA ignores any

digits from the caller if bargein is set to F. The default value of the bargein attribute is F.

In a <form> dialog, if <audio> dialog is not been defined, AA does not play a prompt. If it is defined,

AA first plays the specified prompt, then executes the action instruction described in the next section.

Action Instruction

The action includes:

•

goto—AA transfers the caller from one dialog state to the other dialog. All dialogs are identified by

the attribute “id”. The value in the id attribute must be unique. Otherwise, AA selects the last valid

dialog as the transfer target dialog.

For example, <goto link= “dir_dlg”>

•

xfer—AA blind transfers the caller to the target and AA ends.

Example: <xfer name= “Technical Support” target= “5000”/>

The “name” attribute is optional. “target” attribute should be a valid target phone number.

exit —When this action is reached, AA is stopped, and the call ends.

For example, </exit>

In one dialog, only one action can be defined. After the xfer or exit action is performed, AA ends

automatically.

Noinput Instruction

The following is an example of the noinput instruction:

<!—audio instruction (optional) -->

</noinput>

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Chapter 5 Configuring the LVS Auto-Attendant

XML Scripting for the Auto-Attendant

The <noinput> dialog can only be used in the menu dialog and is optional. When it is specified, AA

executes the audio and action instructions if the user does not input any digits with the value of the

<timeout> parameter, in seconds. If the repeat attribute is set to T, AA plays the menu prompt after

playing the prompt specified in the <noinput> dialog and ignores the action instruction. If the value is

F, AA executes the action instruction. The default value of the repeat attribute is F.

Either the audio or the action instruction can be empty. If both are empty, AA does nothing and waits

for user input.

Nomatch Instruction

The following is an example of the nomatch instruction:

<!—audio instruction (optional) -->

</nomatch>

The <nomatch> dialog can be used only in a menu dialog and is optional. This dialog is activated when

DMTF digits do not match the dial plan. When the nomatch condition is met, AA executes the audio

and action instructions in the <nomatch> dialog. If the repeat attribute is set to T, AA plays the menu

prompt after playing the no input prompt and ignores the action instruction. If the repeat attribute is set

to F, AA executes the action instruction. The default value of the repeat attribute is F

Either the audio or action instruction can be empty. If both are empty, AA does nothing and ignores all

buffered digits.

Menu Matched Instruction—Recognition of Touch Tone (DMTP) Key Presses

The following is an example of the match instruction:

<!—audio instruction (optional) -->

</case>

<!—audio instruction (optional) -->

</case>

<!—audio instruction (optional) -->

</default>

</match>

The <match> dialog can be used only in the menu dialog and it is a mandatory field. When the DTMF

digits match the dialplan, the <match> dialog is activated. AA compares each <case> dialog and

executes the corresponding audio/action instructions. If AA cannot find a match in any <case> dialogs,

it performs the <default> dialog audio/action instruction if <default> is defined; otherwise, AA ends.

The user can specify exact numbers, (for example 1, 23, 1234 and so on), in the input attribute of the

<case> dialog, or the user can use the dial pattern (for example, “50xx”, “408xxx5061”, “xx.”). The user

can also combine several dial patterns together and use “|” to separate them (for example,

“50xx|408xxx5061|1234”).

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Chapter 5 Configuring the LVS Auto-Attendant

AA XML Script Examples

The user can use the variable “$input” in the target attribute of the <xfer> action. The value of this

variable means that the input value that is already passed by the dialplan. AA does no translation, but

directly transfers the call to the target.

For example:

</default>

Currently, only outbound DTMF (INFO/AVT) can be recognized by AA.

AA XML Script Examples

This section provides examples of XML scripts for configuring AA. It includes the following topics:

•

Example 1—AA Default XML Script, page 5-17

Example 2, page 5-18

Example 3—AA Script with Two Treatments, page 5-19

Example 1—AA Default XML Script

In Example 1, when there is an incoming call, AA picks up the call in four seconds. AA then plays

prompt1. The caller can interrupt the prompt at any time. This means that at the same time, AA is ready

to accept DTMF inputs. After the prompt is finished playing, the no-input timer is turned on. If the caller

does not input any digits in 10 seconds, the no-input dialog is executed.

AA repeats prompt1 and waits for user DTMF inputs. If the caller inputs digits within 10 seconds, but

it does not match the dial plan defined in <AA Dial plan 1>, the no-match dialog is executed. It plays

prompt3 and waits for user DTMF input. Otherwise, the <match> dialog is executed. AA plays prompt2

and the user is transferred to the target extension.

Exam ple 5-1 AA Script—Exam ple 1

<AA Answer Delay> = “4” (in seconds)

<AA Dial Plan 1> = “(<1:5001>|<2:5002>|500x|xxxxx.)”

<AA Script> =

“<aa>

<match>

</default>

</match>

</form>

</aa>”

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Chapter 5 Configuring the LVS Auto-Attendant

AA XML Script Examples

Example 2

In Example 2, when there is an incoming call to AA, AA picks up the call in eight seconds. AA then

plays prompt1. The caller can interrupt the prompt at any time. If the caller inputs digits within 10

seconds, but it does not match the dial plan defined in <AA Dial plan 1>, the nomatch dialog is executed.

It plays prompt3 and wait for DTMF input. Otherwise, the <match> dialog is executed.

If the DTMF input is “1”, AA goes to the “Sales” form. Otherwise, AA plays prompt2 and the user is

transferred to the target extension.

In the “Sales” form, AA plays prompt5 and waits for DTMF inputs. <AA Dial plan 2> is used as the

form 2 menu dialplan. When the user enters a star (*), prompt 4 is played, AA exits, and the call ends.

When the user enters 0, 1, or 2, the user is transferred to extension 5000, 5001, or 5002, respectively.

Exam ple 5-2 AA Script—Exam ple 2

<AA Answer Delay> = “8” (in seconds)

<AA Dial Plan 1> = “(1|7xxx)” (valid office extension: 4 digits and starts with 7)

<AA Dial Plan 2> = “(*|<1:5001>|<2:5002>|<0:5000>)”

<AA Script> =

“<aa>

<match>

</case>

</default>

</match>

</form>

<match>

<exit/>

</case>

</default>

</match>

</form>

</aa>”

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Chapter 5 Configuring the LVS Auto-Attendant

AA XML Script Examples

In this example, the following messages are recorded using IVR:

Table 5-4 Prom pts for Exam ple 2

Prompt ID

Message

Prompt1

“Welcome to ABC company. For Sales, enter 1. If you know your party’s

extension, you may enter it now.”

Prompt2

Prompt3

Prompt4

Prompt5

“Your call has been transferred”

“Not a valid extension, please try again”

“Goodbye”

“Press 1 for price info, press 2 for return, press 0 for sales representative,

press * to exit”

Example 3—AA Script with Two Treatments

This example shows an AA script with two treatments for different time periods:

Office Hour AA Treatment, page 5-19

Non-Office Hour AA Treatment, page 5-20

The details of this scenario are as follows:

•

Travel agency Company: “All Seasons”

Company lobby representative extension: 501

Sales Group extension: 200

Support Group extension: 300

Holiday Getaway Special representative extension: 202

Valid employee extensions: 5xx or 8001

Office Hour AA Treatment

In the Office Hour treatment in Example 3, an incoming call on Line 1 rings Extension 501 (receptionist)

and AA at same time. If the receptionist does not pick up the call in 8 seconds, the call is connected to

AA. AA announces prompts 5, 8, and 9 in sequence, and then waits up to 15 seconds for user input. If

no input is received, then the call is forwarded to Extension 501. The following summarizes the system

response to different user inputs:

•

An invalid extension: Prompt 3 is played and AA continues waiting for DTMF inputs.

1: Prompts 7, 8, and then 9 are played and AA continues waiting for DTMF inputs.

2: Prompt 2 is played and AA begins executing the SUPPORT form and starts playing prompt 10.

The dial plan has been changed to the AA Dial Plan 2 parameter.

Form SUPPORT menu execution: AA waits up to 10 seconds for user input. If no input is received,

then the call is forwarded to 200 (Support group line). Upon matched inputs:

•

* (asterisk): Auto-Attendant resumes with the “dt”.

1: prompt2 is played and AA transfers the call to 202(Holiday Getaway Special line)

200: prompt2 is played and the AA transfers the call to 200 (Support group line)

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Chapter 5 Configuring the LVS Auto-Attendant

AA XML Script Examples

•

All other valid extensions (defined here to be 0, 3, 5xx, 8001) are forwarded to the associated client.

0 is replaced with 501 (receptionist), 3 is replaced with 300 (Sales group line); 5xx and 8001 are

valid extensions.

Non-Office Hour AA Treatment

In the Non-Office Hour Treatment in Example 3, an incoming call on Line 1 rings Extension 501

(receptionist) and AA at same time. If the receptionist does not pick up the call in four seconds, the call

is connected to AA. AA announces prompts 6, 7, and 8 in sequence, and then waits up for user input.

Prompts 6, 7, 8 are played every 10 seconds, until the user inputs an extension. Only dialplan (501|8001)

is applied to this form. When matched, the user is transferred to the corresponding extension.

Exam ple 5-3 Auto-Attendant Script w ith Tw o Treatm ents

<AA Script1> =

“<aa>

</noinput>

<match>

</case>

</case>

</default>

</match>

</form>

</noinput>

<match>

</case>

</default>

</match>

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Chapter 5 Configuring the LVS Auto-Attendant

AA XML Script Examples

</form>

</aa>”

<AA Script2> =

“<aa>

</nomatch>

<match>

</default>

</match>

</form>

</aa>”

In Example 3, the following messages are recorded through the IVR:

Table 5-5 Prom pts for Exam ple 3

Prompt ID

Prompt 1-4

Prompt5

Prompt6

Prompt7

Message

See the defaults in Table 3: Default Auto-Attendant Prompts.

“Welcome to All Seasons Travel.”

“Thank you for calling All Seasons Travel. Presently we are closed.”

“We are open Monday through Friday 9 AM to 6 PM, and we are closed on

Saturdays and Sundays. Our address is 101 Main Street, Anytown, Anystate,

USA.”

Prompt8

Prompt9

“If you know your party’s three-digit extension, you may enter it now.”

“To reach our receptionist, press 0 at any time. For our company location,

press 1. For travel support, press 2. For sales, press 3. Otherwise, please stay

on the line for our receptionist.”

Prompt10

“Welcome to the All Seasons Travel support line. If this is regarding our

Holiday Getaway Special, please press 1; otherwise, please stay on the line

for one of our travel associates. Or press * to go back.”

The following table summarizes the parameters for Example 3:

Table 5-6 Param eters for Exam ple 3

Parameter Name

(Line 1) Contact List

AA Dial Plan 1

AA Dial Plan 2

AA script 1

Parameter Value

501,aa

(<0:501>|1|2|<3:300>|5xx|8001)

(*|<1:202>|200)

See <AA script1> sample above

See <AA script2> sample above

AA script 2

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Chapter 5 Configuring the LVS Auto-Attendant

Auto-Attendant XML Instructions Set

Table 5-6 Param eters for Exam ple 3

Parameter Name

DayTime AA

Parameter Value

Yes

DayTime AA Script

DayTime Answer Delay 8

NightTime AA

Yes

NightTime AA Script

NightTime Answer

Delay

Weekend/Holiday AA

Yes

Weekend/Holiday AA

Script

Weekend/Holiday

Answer Delay

Day Time:

Start=9:0:0;end=18:0:0

Weekends/Holidays

wk=6,7;

Auto-Attendant XML Instructions Set

The complete set of XML instructions are described in the following table:

Instruction

Description

Syntax and Example(s)

dialplan

This determines the dialplan id of the current menu

<form>. AA processes the user input according to the

dial plan and then is dispatched to the match, nomatch,

or noinput instruction.

“dp1” matches the AA Dial Plan 1 parameter

found on the Voice - SIP screen of the

administration web server.

“dp2” matches the AA Dial Plan 2 parameter

found on the Voice - SIP screen of the

administration web server.

</noinput>

noinput

When specified, AA executes the specified audio and

action instructions if the user does not input any digits

in <timeout> seconds. If the repeat attribute is set to

“T”, then AA plays the menu prompt after playing the

prompt specified in the <noinput> audio instruction

and ignore the action instruction; otherwise, AA

executes the action instruction. By default, “repeat” is

“F”.

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Auto-Attendant XML Instructions Set

Instruction

Description

Syntax and Example(s)

nomatch

When specified, the nomatch instruction runs when the

user input digits do not match anything in the dial plan.

AAAA executes the specified audio and action

instructions. If the repeat attribute is set to “T”, AA

plays the menu prompt after playing the no input

prompt and ignore the action instruction; otherwise,

AA executes the action instruction. By default,

“repeat” is “F”.

</nomatch>

</case>

match

Upon a match between the user input and the dial plan,

AA transfers to the corresponding <case> and execute

the corresponding audio and/or action instructions. If

AA cannot find a match in any of the <case>

statements, it performs the <default> case.

<!—audio instruction (optional) -->

</case>

<!—audio instruction (optional) -->

</default>

</match>

Action goto

Instruc

tions

AA transfers the caller from one <form> to the other

<form>. All <form>s are identified by the attribute

“id”. The value in the id attribute must be unique;

otherwise, AA selects the last valid <form> as the

transfer-to target.

“daytime” is the id of a <form> entry.

Example: <form id=”daytime” type=”menu”>

<xfer name= “Technical Support” target=

“5000”/>

xfer

AA performs a blind transfer of the caller to the target,

and then it ends processing “target = $input” is

equivalent to the input value already passed by the

dialplan. There is no significance to the name attribute.

<exit>

exit

When this action is reached, AA stops, and the call

ends

audio

AA plays the audio specified in the “src” attribute. This

attribute must be prompt<n>, with <n> being a number

in the range 1–10. When playing the audio, AA allows

the caller to interrupt the current prompt by pressing

digits if the bargein attribute is set to “T”. AA ignores

any digits from the caller if the bargein attribute is set

to “F” (the default value).

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Auto-Attendant XML Instructions Set

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C H A P T E R

SPA9000 Field Reference

This chapter describes the fields within each section of the following administration web server pages:

•

Info Tab, page 6-2

System Tab, page 6-6

SIP Tab, page 6-7

Regional Tab, page 6-23

FXS 1/2 Tab, page 6-38

Line 1/2/3/4 Tab, page 6-47

For information about the tabs on the Routing page, see the documentation for any Linksys router. For

information about the Provisioning page, see the Linksys SPA Provisioning Guide.

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Chapter 6 SPA9000 Field Reference

Info Tab

This section describes the fields for the following headings on the Info tab:

•

Product Information, page 6-2

System Status, page 6-2

FXS 1/2 Status, page 6-3

Line 1/2/3/4 Status, page 6-3

Auto Attendant Prompt Status, page 6-4

Internal Music Status, page 6-5

Note

The fields on this tab are read-only and cannot be edited.

Product Information

Field

Description

Product Name

Serial Number

Software Version

Hardware Version

MAC Address

Client Certificate

Model number of the SPA9000.

Serial number of the SPA9000.

Version number of the SPA9000 software.

Version number of the SPA9000 hardware.

MAC address of the SPA9000.

Status of the client certificate, which authenticates the SPA9000 for use in the ITSP

network. This field indicates if the client certificate is properly installed in the SPA9000.

Customization

Licenses

For an RC unit, this field indicates whether the unit has been customized or not. Pending

indicates a new RC unit that is ready for provisioning. If the unit has already retrieved its

customized profile, this field displays the name of the company that provisioned the unit.

Indicates any additional licenses that you have installed in the SPA9000.

System Status

Field

Description

Current Time

Elapsed Time

Current date and time of the system; for example, 10/3/2003 16:43:00.

Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.

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Info Tab

Line 1/2/3/4 Status

)

Field

Description

Registration State

Last Registration At

Next Registration In

Message Waiting

Last date and time the line was registered.

Number of seconds before the next registration renewal.

Indicates whether you have new voicemail waiting: Yes or No. This is updated when

voicemail notification is received. You can also manually modify it to clear or set the flag.

Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is

stored in long term memory and survives after reboot or power cycle.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

FXS 1/2 Status

Field

Description

Hook State

Status of phone readiness. On indicates that the phone is ready for use; Off indicates that

the phone is in use.

Message Waiting

Indicates whether you have new voicemail waiting: Yes or No. This is updated when

voicemail notification is received. You can also manually modify it to clear or set the flag.

Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is

stored in long term memory and survives after reboot or power cycle.

Call Back Active

Last Called Number

Last Caller Number

Mapped SIP Port

Call 1/2 State

Indicates whether a call back request is in progress: Yes or No.

Last number called.

Number of the last caller.

Port number of the SIP port mapped by NAT.

Status of the call.

Call 1/2 Tone

Type of tone used by the call.

Call 1/2 Encoder

Call 1/2 Decoder

Call 1/2 FAX

Codec used for encoding.

Codec used for decoding.

Status of the fax pass-through mode.

Direction of the call.

Call 1/2 Type

Call 1/2 Remote Hold

Call 1/2 Callback

Call 1/2 Peer Name

Indicates whether the far end has placed the call on hold.

Indicates whether the call was triggered by a call back request.

Name of the internal phone.

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Chapter 6 SPA9000 Field Reference

Info Tab

Call 1/2 Peer Phone

Call 1/2 Duration

Phone number of the internal phone.

Duration of the call.

Call 1/2 Packets Sent

Call 1/2 Packets Recv

Call 1/2 Bytes Sent

Call 1/2 Bytes Recv

Call 1/2 Decode Latency

Call 1/2 Jitter

Number of packets sent.

Number of packets received.

Number of bytes sent.

Number of bytes received.

Number of milliseconds for decoder latency.

Number of milliseconds for receiver jitter.

Number of milliseconds for delay.

Number of packets lost.

Call 1/2 Round Trip Delay

Call 1/2 Packets Lost

Call 1/2 Packet Error

Number of invalid packets received.

Auto Attendant Prompt Status

Field

Description

Prompt 1

Default. If you change a default, the screen shows the new prompt’s duration in

milliseconds.

Prompt 2

Prompt 3

Prompt 4

Default. If you change a default, the screen shows the new prompt’s duration in

milliseconds.

Default. If you change a default, the screen shows the new prompt’s duration in

milliseconds.

Default. If you change a default, the screen shows the new prompt’s duration in

milliseconds.

Prompt 5

For this prompt, the screen shows its duration in milliseconds.

Number of milliseconds available.

Prompt 6

Prompt 7

Prompt 8

Prompt 9

Prompt 10

Space Remaining

Current AA

Auto-attendant in use.

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Info Tab

Internal Music Status

Field

Description

Installed Music Path

Displays the pathname for the music source used for the music-on-hold feature.

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Chapter 6 SPA9000 Field Reference

System Tab

This section describes the fields for the following headings on the System tab:

•

System Configuration, page 6-6

Miscellaneous Settings, page 6-6

System Configuration

Field

Description

Restricted Access Domains

Enable Web Admin Access

This feature is used when implementing software customization.

Lets you enable or disable local access to the administration web server. Select yes or no

from the drop-down menu.

The default is yes.

Admin Passwd

User Password

Password for the administrator.

The default is no password.

Password for the user.

The default is no password.

Miscellaneous Settings

Field

Description

Syslog Server

Debug Server

Specifies the IP address of the syslog server.

Specifies the IP address of the debug server, which logs debug information. The level of

detailed output depends on the debug level parameter setting.

Debug Level

Determines the level of debug information that is generated. Select 0, 1, 2, or 3 from the

drop-down menu. The higher the debug level, the more debug information is generated.

The default is 0, which indicates that no debug information is generated.

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Chapter 6 SPA9000 Field Reference

SIP Tab

This section describes the fields for the following headings on the SIP tab:

•

SIP Parameters, page 6-7

SIP Timer V a lues (sec), page 6-8

Response Status Code Handling, page 6-10

RTP Parameters, page 6-10

SDP Payload Types, page 6-11

N A T S upport Parameters, page 6-13

PBX Parameters, page 6-14

Internal Music Source Parameters, page 6-17

Auto Attendant Parameters, page 6-18

PBX Phone Parameters, page 6-21

SIP Parameters

Field

Description

Max Forward

SIP Max Forward value, which can range from 1 to 255.

The default is 70.

Max Redirection

Max Auth

Number of times an invite can be redirected to avoid an infinite loop.

The default is 5.

Maximum number of times (from 0 to 255) a request may be challenged.

The default is 2.

SIP User Agent Name

User-Agent header used in outbound requests.

The default is $VERSION. If empty, the header is not included. Macro expansion of $A

to $D corresponding to GPP_A to GPP_D allowed.

SIP Server Name

Server header used in responses to inbound responses.

The default is $VERSION.

SIP Reg User Agent Name

User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User

Agent Name> is also used for the REGISTER request.

The default is blank.

SIP Accept Language

Accept-Language header used. There is no default (this indicates SPA9000 does not

include this header). If empty, the header is not included.

DTMF Relay MIME Type

MIME Type used in a SIP INFO message to signal a DTMF event.

The default is application/dtmf-relay.

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Hook Flash MIME Type

MIME Type used in a SIP INFO message to signal a hook flash event.

The default is application/hook-flash.

Remove Last Reg

Lets you remove the last registration before registering a new one if the value is different.

Select yes or no from the drop-down menu.

The default is no.

Use Compact Header

Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the

drop-down menu. If set to yes, the SPA9000 uses compact SIP headers in outbound SIP

messages. If set to no, the SPA9000 uses normal SIP headers. If inbound SIP requests

contain compact headers, SPA9000 reuses the same compact headers when generating the

response regardless the settings of the <Use Compact Header> parameter. If inbound SIP

requests contain normal headers, SPA9000 substitutes those headers with compact

headers (if defined by RFC 261) if <Use Compact Header> parameter is set to yes.

The default is no.

Escape Display Name

RFC 2543 Call Hold

Lets you keep the Display Name private. Select yes if you want the SPA9000 to enclose

the string (configured in the Display Name) in a pair of double quotes for outbound SIP

messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of

double quotes. Otherwise, select no.

The default is no.

If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to

the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP.

The unit will always include a=sendonly syntax in the SDP in either case.

The default is yes.

SIP TCP Port Min

SIP TCP Port Max

CTI Enable

Specifies the lowest TCP port number that can be used for SIP sessions.

Specifies the highest TCP port number that can be used for SIP sessions.

Enables or disables the Computer Telephone Interface feature provided by some servers.

SIP Timer Values (sec)

Field

Description

SIP T1

RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.

The default is.5.

SIP T2

SIP T4

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE

responses), which can range from 0 to 64 seconds.

The default is 4.

RFC 3261 T4 value (maximum duration a message remains in the network), which can

range from 0 to 64 seconds.

The default is 5.

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SIP Timer B

SIP Timer F

SIP Timer H

SIP Timer D

SIP Timer J

INVITE time-out value, which can range from 0 to 64 seconds.

The default is 32.

Non-INVITE time-out value, which can range from 0 to 64 seconds.

The default is 32.

INVITE final response, time-out value, which can range from 0 to 64 seconds.

The default is 32.

ACK hang-around time, which can range from 0 to 64 seconds.

The default is 32.

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.

The default is 32.

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included

in the request.

The default is 240. Range: 0–(2³¹–1).

ReINVITE request Expires header value. If you enter 0, the Expires header is not

included in the request.

ReINVITE Expires

Reg Min Expires

The default is 30. Range: 0–(2³¹–1).

Minimum registration expiration time allowed from the proxy in the Expires header or as

a Contact header parameter. If the proxy returns a value less than this setting, the

minimum value is used.

The default is 1.

Reg Max Expires

Reg Retry Intvl

Maximum registration expiration time allowed from the proxy in the Min-Expires header.

If the value is larger than this setting, the maximum value is used.

The default is 7200.

Interval to wait before the SPA9000 retries registration after failing during the last

registration.

The default is 30.

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match<Retry Reg RSC>,

the SPA9000 waits for the specified length of time before retrying. If this interval is 0, the

SPA9000 stops trying. This value should be much larger than the Reg Retry Intvl value,

which should not be 0.

The default is 1200.

Reg Retry Random Delay

Random delay range (in seconds) to add to <Register Retry Intvl> when retrying

The default is 0, which disables this feature.

Reg Retry Long Random Delay Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying

REGSITER after a failure. This feature was added in Release 5.1.

The default is 0, which disables this feature.

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Reg Retry Intvl Cap

The maximum value to cap the exponential back-off retry delay (which starts at <Register

Retry Intvl> and doubles on every REGISTER retry after a failure). In other words, the

retry interval is always at <Register Retry Intvl> seconds after a failure. If this feature is

enabled, <Reg Retry Random Delay> is added on top of the exponential back-off adjusted

delay value. This feature was added in Release 5.1.

The default value is 0, which disables the exponential back-off feature.

Response Status Code Handling

Field

Description

SIT1 RSC

SIP response status code for the appropriate Special Information Tone (SIT). For example,

if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is

returned, the SIT1 tone is played. Reorder or Busy Tone is played by default for all

unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.

SIT2 RSC

SIP response status code to INVITE on which to play the SIT2 Tone.

SIP response status code to INVITE on which to play the SIT3 Tone.

SIP response status code to INVITE on which to play the SIT4 Tone.

SIP response code that retries a backup server for the current request.

SIT3 RSC

SIT4 RSC

Try Backup RSC

Retry Reg RSC

Interval to wait before the SPA9000 retries registration after failing during the last

registration.

The default is 30.

RTP Parameters

Field

Description

RTP Port Min

Minimum port number for RTP transmission and reception. <RTP Port Min> and

<RTP Port Max> should define a range that contains at least 4 even number

ports, such as 100 – 106.

The default is 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.

The default is 16482.

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple

of 0.01 seconds.

The default is 0.030.

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Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer

before the SPA9000 terminates the call. If value is set to 0, the SPA9000 ignores the limit

on ICMP errors.

The default is 0.

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection. It can range from

0 to 255 seconds. During an active connection, the SPA9000 can be programmed to send

out compound RTCP packet on the connection. Each compound RTP packet except the

last one contains a SR (Sender Report) and a SDES.(Source Description). The last RTCP

packet contains an additional BYE packet. Each SR except the last one contains exactly

1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME,

and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to

<Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the

Vendor/Hardware-platform-software-version (such as Linksys/SPA9000-1.0.31(b)). The

NTP timestamp used in the SR is a snapshot of the SPA9000’s local time, not the time

reported by an NTP server. If the SPA9000 receives a RR from the peer, it attempts to

compute the round trip delay and show it as the <Call Round Trip Delay> value (ms) in

the Info section of SPA9000 web page.

The default is 0.

No UDP Checksum

Stats In BYE

Select yes if you want the SPA9000 to calculate the UDP header checksum for SIP

messages. Otherwise, select no.

The default is no.

Determines whether the SPA9000 includes the P-RTP-Stat header or response to a BYE

message. The header contains the RTP statistics of the current call. Select yes or no from

the drop-down menu. The format of the P-RTP-Stat header is:

P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets

received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration in

s>,EN=<encoder>,DE=<decoder>.

The default is no.

SDP Payload Types

The configured dynamic payloads are used for outbound calls only where the SPA9000 presents the SDP

offer. For inbound calls with a SDP offer, SPA9000 follow the caller dynamic payload type assignments.

SPA9000 uses the configured codec names in its outbound SDP. SPA9000 ignores the codec names in

incoming SDP for standard payload types (0 – 95). For dynamic payload types, SPA9000 identifies the

codec by the configured codec names. Comparison is case-insensitive.

Field

Description

NSE Dynamic Payload

NSE dynamic payload type. The valid range is 96-127.

The default is 100.

AVT Dynamic Payload

AVT dynamic payload type. The valid range is 96-127.

The default is 101.

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INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

There is no default.

G726r16 Dynamic Payload

G726r24 Dynamic Payload

G726r40 Dynamic Payload

G729b Dynamic Payload

NSE Codec Name

G.726-16 dynamic payload type. The valid range is 96-127.

The default is 98.

G.726-24 dynamic payload type. The valid range is 96-127.

The default is 97.

G.726-40 dynamic payload type. The valid range is 96-127.

The default is 96.

G.729b dynamic payload type. The valid range is 96-127.

The default is 99.

NSE codec name used in SDP.

The default is NSE.

AVT Codec Name

AVT codec name used in SDP.

The default is telephone-event.

G711u Codec Name

G.711u codec name used in SDP.

The default is PCMU.

G711a Codec Name

G.711a codec name used in SDP.

The default is PCMA.

G726r16 Codec Name

G726r24 Codec Name

G726r32 Codec Name

G726r40 Codec Name

G729a Codec Name

G.726-16 codec name used in SDP.

The default is G726-16.

G.726-24 codec name used in SDP.

The default is G726-24.

G.726-32 codec name used in SDP.

The default is G726-32.

G.726-40 codec name used in SDP.

The default is G726-40.

G.729a codec name used in SDP.

The default is G729a.

G729b Codec Name

G723 Codec Name

G.729b codec name used in SDP.

The default is G729ab.

G.723 codec name used in SDP.

The default is G723.

EncapRTP Codec Name

EncapRTP codec name used in SDP.

The default is EncapRTP.

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NAT Support Parameters

Field

Description

Handle VIA received

If you select yes, the SPA9000 processes the received parameter in the VIA header (this

is inserted by the server in a response to anyone of its requests). If you select no, the

parameter is ignored. Select yes or no from the drop-down menu.

The default is no.

Handle VIA rport

If you select yes, the SPA9000 processes the rport parameter in the VIA header (this is

inserted by the server in a response to anyone of its requests). If you select no, the

parameter is ignored. Select yes or no from the drop-down menu.

The default is no.

Insert VIA received

Insert VIA rport

Inserts the received parameter into the VIA header of SIP responses if the received-from

IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.

The default is no.

Inserts the rport parameter into the VIA header of SIP responses if the received-from IP

and VIA sent-by IP values differ. Select yes or no from the drop-down menu.

The default is no.

Substitute VIA Addr

Send Resp To Src Port

STUN Enable

Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the

drop-down menu.

The default is no.

Sends responses to the request source port instead of the VIA sent-by port. Select yes or

no from the drop-down menu.

The default is no.

Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down

menu.

The default is no.

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the SPA9000

can perform a NAT-type discovery operation when it powers on. It contacts the configured

STUN server, and the result of the discovery is reported in a Warning header in all

subsequent REGISTER requests. If the SPA9000 detects symmetric NAT or a symmetric

firewall, NAT mapping is disabled.

The default is no.

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT

mapping discovery.

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EXT IP

External IP address to substitute for the actual IP address of the SPA9000 in all outgoing

SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the SPA9000 assumes this IP address when generating SIP

messages and SDP (if NAT Mapping is enabled for that line). However, the results of

STUN and VIA received parameter processing, if available, supersede this statically

configured value.

The default is 0.0.0.0.

EXT RTP Port Min

External port mapping number of the RTP Port Min. number. If this value is not zero, the

RTP port number in all outgoing SIP messages is substituted for the corresponding port

value in the external RTP port range.

The default is 0.

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

The default is 15.

PBX Parameters

Field

Description

Proxy Network Interface

This setting tells the SPA9000 how the client stations are connected. Choices: {LAN,

WAN}. Default is WAN. The SPA9000 communicates with client stations via the selected

interface only.

Proxy Listen Port

Multicast Address

Group Page Address

Port at which the SPA9000 listens for client messages at the selected network interface.

The proxy also sends SIP messages from this port.

The default is 6060.

IP address (and port number) where the SPA9000 sends control messages to all the client

stations at once. This must be a multicast address and must contain a port number.

The default is 224.168.168.168:6061.

IP address (and port number) where the SPA9000 tells the client stations to send and

receive group page RTP packets. This must be a multicast address and must contain a port

number.

The default is 224.168.168.168:34567.

Max Expires

Sets the maximum allowed Registration expires value in seconds for client stations.

Linksys recommends using a smaller value, such as 60 or 120.

The default is 3600.

Force Media Proxy

Forces external client stations to use the SPA9000 Media Proxy when exchanging RTP

traffic with external peers. Linksys recommends using a smaller value, such as 60 or 120.

The default is no.

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Proxy Debug Option

Controls what SIP messages to log that are received at or sent from the Proxy listen port.

Choices are as follows: {

•

none—No logging.

1-line—Logs the start-line only for all messages,

1-line excl. OPT—Same as 1-line but excludes OPTIONS request/response.

1-line excl. NTFY—Same as 1-line but excludes NOTIFY request/response.

1-line excl. REG—Same as 1-line but excludes REGISTER request/response.

1-line excl. OPT|NTFY|REG—Same as 1-line but excludes OPTIONS, NOTIFY, and

•

full—Logs all SIP messages in full text.

full excl. OPT—Same as full but excludes OPTIONS request/response.

full excl. NTFY—Same as full but excludes NOTIFY request/response.

full excl. REG—Same as full but excludes REGISTER request/response.

full excl. OPT|NTFY|REG—Same as full but excludes OPTIONS, NOTIFY, and

The default is is none.

Call Routing Rule

Special dial plan that determines which line interfaces can be used for an external

outbound call request from client station based solely on the target public number. The

dial plan is in the (rule|rule|…|rule) format where:

•

rule = <:Ln[,n[,n[,n]]]>pattern

n = 1, 2, 3, or 4,

pattern = any digit pattern (see <Dial Plan> on how to choose a digit pattern).

If the target number matches the pattern of a rule, the Line indices in the rule’s prefix are

the line interfaces that can be used to make that call. Matches are performed from left to

right, so make sure the most specific rules are placed first. For example:

The default is is (<:L1,2,3,4>9xx.), which specifies that any of the four line interfaces can

be used for any target number starting with 9.

Call Park MOH Server

Specifies the MOH Server to be used to handle a parked call. For example:

[email protected]:5082.

If this parameter is not specified, the internal parking lot is used to host the parked call,

in which case the parked caller hears the internal music file.

The default is is imusic.

Call Park DLG Refresh Intvl

Group 1/2/3/4 User ID

Specifies the interval in seconds between refreshing a call park session. Default is 0,

which disables session refreshes.

Comma-separated list of user-id patterns. A client station whose user-id matches any of

the give patterns is considered as belonging to that group. If the user-id matches more than

one group, the smallest group number is assumed. If the user-id does not match any group,

the client station is considered as belonging to the default group (also known as Group 0).

Each user-id pattern allows * and ? wildcards as well as %xx escaped characters.

The default is is blank, so all client stations belong to the default group.

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Group 1/2/3/4 Line

Ordered comma-separated list of Line indices. SPA9000 attempts to make external calls

for client stations belonging to the group in the order according to this list. Example: 1,3.

The default is is blank, which means that no external calls are allowed for client stations

in this group.

Default Group Line

Hunt Groups

Same as <Group 1/2/3/4 Line>, but applies to the default group.

The default is is 1,2,3,4.

Defines one or more hunt groups that can be called directly by any client station like a

regular extension. The syntax is the same as <Contact List>. Each defined group

extension and name also appears in the corporate directory. This parameter is parsed twice

by the SPA9000 such that a group member of one group can also be the extension of

another group (that is, one level of recursion allowed).

SIP DIDN Field

Determines which field is used to indicate the DID number for an incoming INVITE to a

line interface. The choices are:

•

TO UserID—The user-id field of the TO header

TO Param—A parameter in the TO header with the name specified in <SIP DIDN

Param Name>, such as didn=1234

SIP DIDN Param Name

Accept All MWI as Line

Parameter name to indicate the DID number in an incoming INVITE message.

The default is is didn.

Choose the line. Choices are 1, 2, 3, 4, or Current. The default is Current.

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Internal Music Source Parameters

Field

Description

Internal Music URL

URL from which to download a music file to be used by the parking lot. The format is:

[tftp://]server-ip-addr[:port]/path

TFTP is the only protocol supported for music download. Default port is 69. Changing

the value of this parameter from the web interface triggers a graceful reboot of the

SPA9000. If a valid entry is specified, the SPA9000 attempts to download the file on

bootup and store the samples in flash memory. The SPA9000 remembers the link where

the stored file is downloaded and does not try to download again on the next reboot.

The music samples are encoded in G711u format at 8000 samples/second. The file should

not contain any extra header information. Maximum length of the file is 65.536 seconds

(524288 bytes).

Internal Music Script

Script that tells the SPA9000 how to play the downloaded music file, in the format

[section[,section[,…]]], where:

•

section = [[n](start/end[/pause])][pause2]

n = number of times to repeat the section before moving to the next section. Default

is 1.

•

start/end = starting and 1+ending sample for this section; note that samples are

numbered from 0 to total-length – 1. You may enter –1 or a very large number if the

end of the file is intended as the ending sample. Default start is 0, and default end is

end of the file.

•

pause = number of samples to pause after the ending sample is played. Default is 0.

pause2 = additional number of samples to pause after the entire n repetitions of the

section are played. Default is 0.

A maximum of 16 sections can be specified. Samples should be encoded in G711u format

at 8000 samples/second. When all sections are played, the SPA9000 replays from the first

section again.

Examples:

•

40000 (plays the entire file, pauses for 5s, then repeats)

2(0/32000),3(32000/100000/4000)2000,(100000/-1)80000

Internal Music Refresh Intvl

Internal Music LBR Codec

Specifies the interval in seconds between refreshing an internal music session.

The default is is 0, which disables session refreshes.

Selects one low bit-rate codec as an alternative to G711u and G711a for playing internal

music. Choices are {none, G729a, G726-16, G726-24, G726-32, G726-40}.

The default is is none.

Internal Music Preferred Codec Selects which codec is the preferred choice to play internal music. Choices are {G711u,

G711a, Low Bit Rate}. Low Bit Rate refers to the selected <Internal Music LBR Codec>.

If <Internal Music LBR Codec> is none, G711u is the preferred codec also.

The default is is G711u.

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Internal Music Use Pref Codec

Only

Forces the internal music player to use the preferred codec only.

The default is is no.

Auto Attendant Parameters

Field

Description

AA Dial Plan 1

Dial Plan 1 to be used in an AA script. Example: <dialplan src=dp1/>.

The default is: (10x|xxx.)

AA Dial Plan 2

AA script 1/2/3

Dial Plan 2 to be used in an AA script. Example: <dialplan src=dp2/>.

The default is: (<:10>x|xxx.)

AA script. See Chapter 4, Configuring the LVS Auto-Attendant, for the complete syntax.

The default is:

</nomatch>

<match>

</default></match>

</form></aa>

Daytime AA

To enable the daytime Auto-Attendant, select yes. Otherwise, select no.

The default is is yes.

Nighttime AA

Weekend/Holiday AA

Day Time

To enable the nighttime Auto-Attendant, select yes. Otherwise, select no.

The default is is no.

To enable this Auto-Attendant, select yes. Otherwise, select no.

The default is is no.

Daytime hours for the daytime Auto-Attendant in 24-hour format. Enter the start and

end times in this format: start=hh:mm:ss;end=hh:mm:ss (hh for hours, mm for

minutes, and ss for seconds).

For example, start=9:0:0;end=17:0:0 means that the start time is 9 AM and the end

time is 5 PM. The other hours (5 PM to 9 AM) are considered nighttime hours.

If you do not enter start and end times, the whole day (24 hours) is considered as daytime,

so the nighttime Auto-Attendant is not used, even if it is enabled.

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DayTime AA Script

Specifies which AA script (1, 2, or 3) is used for the AA treatment when operating in

daytime mode.

The default is 1.

NightTime AA Script

Weekend/Holiday AA Script

Weekends/Holidays

Specifies which AA script (1, 2, or 3) is used for the AA treatment when operating in

daytime mode.

The default is 1.

Specifies which AA script (1, 2, or 3) is used for the AA treatment when operating in

daytime mode.

The default is 1.

When the weekend/holiday Auto-Attendant is enabled, you can use this setting to

specify the weekends and holidays. Up to four weekend days can be defined. Use

this format:

[wk=n1[,ni];][hd=mm/dd/yyyy|mm/dd/yyyy-mm/dd/yyyy[,mm/dd/yyyy|mm/dd/yyy

y-mm/dd/yyyy];]

•

(wk for weekend, which can be 1 for Monday to 7 for Sunday)

•

(hd for holiday, which does not have to include the year)

For example, wk=6,7;hd=1/1,2/21/2006,5/30/2006,12/19/2006-12/30/2006 means that

Saturdays and Sundays are the weekends. Holidays are January 1-2, 2006; May 30, 2006;

and December 19-30, 2006.

DayTime Answer Delay

NightTime Answer Delay

Number of seconds before the AA answers when operating in the daytime mode.

The default is 12.

Number of seconds before the AA answers when operating in the nighttime mode.

The default is 0.

Weekend/Holiday Answer

Delay

Number of seconds before the AA answers when operating in the weekend/holiday mode.

The default is 0.

AA LBR Codec

Selects one low bit rate codec as an alternative to G711u and G711a for playing AA

prompts. Choices are {none, G729a, G726-16, G726-24, G726-32, G726-40}.

The default is none.

AA Preferred Codec

Selects which codec is the preferred choice to play AA prompts. Choices are {G711u,

G711a, Low Bit Rate}. Low Bit Rate refers to the selected <AA LBR Codec>. If <AA

LBR Codec> is none, G711u is the preferred codec also.

The default is G711u.

AA User Pref Codec Only

Forces the AA to use the preferred codec only.

The default is no.

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SIP Tab

AA Prompts URL Script

Instructs the SPA9000 to erase or download user-recorded prompt files from a

TFTP/HTTP/HTTPS server. These files must be encoded in G711u, size less than 60

seconds, with the header removed.

The sum of the prompt files cannot be longer than 94.5 seconds. The prompt is

downloaded when the device boots. If the prompt has already been downloaded from the

given URL, the download does not occur. If prompt file name is none, the corresponding

prompt currently saved in the flash is erased. The default value is blank. This parameter

was added in release 5.1.The following is the format of the prompt file:

serv=scheme://server_addr[:port]/root_path;[p1={prompt1 file path

name};][p2={prompt2 file path name};][p3={prompt3 file path

name};][p4={prompt4 file path name};][p5={prompt5 file path

name};][p6={prompt6 file path name};][p7={prompt7 file path

name};][p8={prompt8 file path name};][p9={prompt9 file path

name};][p10={prompt10 file path name};]

Where:

•

scheme = tftp|http|https

default port is 69 for tftp, 80 for http, and 443 for https

root_path can be empty

[ ] denotes optional item

For example:

serv=tftp://192.168.2.150/root/test/;p1=menu.wav;p2=transfer.wav;p3=nomatch.w

av;p4=none;

The following shows the source for each prompt in this example:

prompt 1: tftp://192.168.2.150/root/test/menu.wav

prompt 2: tftp://192.168.2.150/root/test/transfer.wav

prompt 3: tftp://192.168.2.150/root/test/nomatch.wav

prompt 4 is erased

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PBX Phone Parameters

Field

Description

Next Auto User ID

User-id assigned to the next (unprovisioned) client station that requests initial profile from

the SPA9000 internal configuration server. The parameter is automatically incremented

once a new user-id is assigned.

The default is is 5000.

Phone Ext Password

Phone Upgrade Rule

A REGISTRATION password to apply on Ext 1 of all the client stations. If password is

not specified, all stations are allowed to register without being challenged by the

SPA9000.

The default is blank.

Upgrade rule for all the client stations. For example:

tftp://192.168.2.207/$PN.bin

Note that the $PN macro is expanded to the product name of the client requesting the

firmware. This allows upgrading phone clients with different firmware using a single rule.

The default is blank.

Phone Dial Plan

Dial plan for the client stations.

The default is

(9,[3469]11S0|9,<:1408>[2-9]XXXXXX|9,<:1>[2-9]xxxxxxxxxS0|9,1[2-9]xxxxxxxxxS

0|9,011xx.|9,xx.|[1-8]xxx).

This dial plan tells the phone to do the following:

•

play the outside dial tone if the first digit is 9

dial 9311, 9411, 9611, and 9911 immediately

dial 9 + [2-9] + 6 digits after a short timeout and insert the 1 + 408 area code

dial 9 + [2-9] + 9 digits immediately and insert the 1 (domestic long distance)

dial 91 + [2-9] + 9 digits immediately (domestic long distance)

dial 9011 + 1 or more digits after timeout or the # (pound) key (international)

dial 9 + 1 or more digits after timeout or the # (pound) key (catch-all)

dial [1-8] + 3 more digits immediately (internal calls)

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Phone Config XML

XML configuration to be served to the phones when they request the init_$MA.xml file

from the SPA9000 during their first-time initialization (either a brand-new phone or after

a factory reset). If this parameter is empty, the SPA9000 serves the normal auto-generated

startup XML configuration file to the phone instead.

The purpose of this parameter is to facilitate ITSP provisioning of new phones remotely.

ITSP can simply include a default phone profile rule parameter in this parameter so that

the phone can obtain it from the SPA9000 on initial power-up. For example:

<Profile_Rule>https://www.itsp.com/init/spa$MA.cfg</Profile_Rule>

Do not include <flat-profile>…</flat-profile>. The SPA9000 automatically adds them

when serving the configuration file to the phones.

The default is blank.

Use LVS_PROXY

If this option is yes, the SPA9000 uses the hostname LVS_PROXY instead of its IP

address in the Profile_Rule parameter that is served to the phones when they request

init_$MA.xml during first-time initialization (when <Phone Config XML> is not

specified). On reboot, the phones resolve the LVS_PROXY by querying the LAN via

multicast. The SPA9000 replies to the query with its actual IP address. This allows the

SPA9000 to use a dynamically-assigned IP address that is not fixed.

Make sure that the phones have a compatible firmware that understands that

LVS_PROXY is a special hostname. For SPA-941/942/921/922/901, use 4.1.12 or later;

for SPA-841, use 3.1.6(KS) or later.

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Chapter 6 SPA9000 Field Reference

Regional Tab

This section describes the fields for the following headings on the Regional tab:

•

Call Progress Tones, page 6-23

Distinctive Ring Patterns, page 6-24

Distinctive Call Waiting Tone Patterns, page 6-26

Distinctive Ring/CWT Pattern Names, page 6-26

Ring and Call Waiting Tone Spec, page 6-27

Control Timer V a lues (sec), page 6-28

V e rtical Service Activation Codes, page 6-29

V e rtical Service Announcement Codes, page 6-33

Outbound Call Codec Selection Codes, page 6-33

Miscellaneous, page 6-35

Call Progress Tones

Field

Description

Dial Tone

Prompts the user to enter a phone number. Reorder Tone is played automatically when

<Dial Tone> or any of its alternatives times out.

The default is 350@-19,440@-19;10(*/0/1+2).

Second Dial Tone

Outside Dial Tone

Alternative to the Dial Tone when the user dials a three-way call.

The default is 420@-19,520@-19;10(*/0/1+2).

Alternative to the Dial Tone. It prompts the user to enter an external phone number, as

opposed to an internal extension. It is triggered by a, (comma) character encountered in

the dial plan.

The default is 420@-19;10(*/0/1).

Prompt Tone

Busy Tone

Prompts the user to enter a call forwarding phone number.

The default is 520@-19,620@-19;10(*/0/1+2).

Played when a 486 RSC is received for an outbound call.

The default is 480@-19,620@-19;10(.5/.5/1+2).

Reorder Tone

Played when an outbound call has failed or after the far end hangs up during an established

call. Reorder Tone is played automatically when <Dial Tone> or any of its alternatives

times out.

The default is 480@-19,620@-19;10(.25/.25/1+2).

Off Hook Warning Tone

Played when the caller has not properly placed the handset on the cradle. Off Hook

Warning Tone is played when Reorder Tone times out.

The default is 480@10,620@0;10(.125/.125/1+2).

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Regional Tab

Ring Back Tone

Played during an outbound call when the far end is ringing.

The default is 440@-19,480@-19;*(2/4/1+2).

Confirm Tone

SIT1 Tone

Brief tone to notify the user that the last input value has been accepted.

The default is 600@-16; 1(.25/.25/1).

Alternative to the Reorder Tone played when an error occurs as a caller makes an

outbound call. The RSC to trigger this tone is configurable on the SIP screen.

The default is 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0).

SIT2 Tone

SIT3 Tone

SIT4 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an

outbound call. The RSC to trigger this tone is configurable on the SIP screen.

The default is 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0).

Alternative to the Reorder Tone played when an error occurs as a caller makes an

outbound call. The RSC to trigger this tone is configurable on the SIP screen.

The default is 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

This is an alternative to the Reorder Tone played when an error occurs as a caller makes

an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

The default is 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0).

MWI Dial Tone

Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.

The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).

Cfwd Dial Tone

Played when all calls are forwarded.

The default is 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2).

Holding Tone

Informs the local caller that the far end has placed the call on hold.

The default is 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1).

Conference Tone

Secure Call Indication Tone

Played to all parties when a three-way conference call is in progress.

The default is 350@-19;20(.1/.1/1,.1/9.7/1).

Played when a call has been successfully switched to secure mode. It should be played

only for a short while (less than 30 seconds) and at a reduced level (less than -19 dBm) so

it does not interfere with the conversation.

The default is 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).

Feature Invocation Tone

Played when a feature is implemented. (Not in PAP2T)

The default is 350@-16;*(.1/.1/1).

Distinctive Ring Patterns

Field

Description

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Ring1 Cadence

Cadence script for distinctive ring 1.

The default is 60(2/4).

Ring2 Cadence

Ring3 Cadence

Cadence script for distinctive ring 2.

The default is 60(.3/.2, 1/.2,.3/4.

Cadence script for distinctive ring 3.

The default is 60(.8/.4,.8/4).

Ring4 Cadence

Ring5 Cadence

Ring6 Cadence

Cadence script for distinctive ring 4.

The default is 60(.4/.2,.3/.2,.8/4).

Cadence script for distinctive ring 5.

The default is 60(.4/.2,.3/.2,.8/4).

Cadence script for distinctive ring 6.

The default is 60(.4/.2,.3/.2,.8/4).

Ring7 Cadence

Ring8 Cadence

Cadence script for distinctive ring 7.

The default is 60(.4/.2,.3/.2,.8/4).

Cadence script for distinctive ring 8.

The default is 60(.4/.2,.3/.2,.8/4).

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Distinctive Call Waiting Tone Patterns

Field

Description

CWT1 Cadence

Cadence script for distinctive CWT 1.

The default is 30(.3/9.7).

CWT2 Cadence

CWT3 Cadence

CWT4 Cadence

CWT5 Cadence

CWT6 Cadence

CWT7 Cadence

CWT8 Cadence

Cadence script for distinctive CWT 2.

The default is 30(.1/.1, .1/9.7).

Cadence script for distinctive CWT 3.

The default is 30(.1/.1, .1/.1, .1/9.5).

Cadence script for distinctive CWT 4.

The default is 30(.1/.1, .3/.1, .1/9.3).

Cadence script for distinctive CWT 5.

The default is 30(.3/.1,.1/.1,.3/9.1).

Cadence script for distinctive CWT 6.

The default is 30(.1/.1, .3/.1, .1/9.3).

Cadence script for distinctive CWT 7.

The default is 30(.1/.1, .3/.1, .1/9.3).

Cadence script for distinctive CWT 8.

The default is 2.3(..3/2).

Distinctive Ring/CWT Pattern Names

Field

Description

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound

call.

The default is Bellcore-r1.

Ring2 Name

Ring3 Name

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound

call.

The default is Bellcore-r2.

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound

call.

The default is Bellcore-r3.

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound

call.

The default is Bellcore-r4.

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Ring5 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound

call.

The default is Bellcore-r5.

Ring6 Name

Ring7 Name

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound

call.

The default is Bellcore-r6.

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound

call.

The default is Bellcore-r7.

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound

call.

The default is Bellcore-r8.

Ring and Call Waiting Tone Spec

Field

Description

Ring Waveform

Waveform for the ringing signal.

The default is Sinusoid.

Ring Frequency

Ring Voltage

Frequency of the ringing signal. Valid values are 10–100 (Hz).

The default is 25.

Ringing voltage. 60–90 (V).

The default is 70.

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.

The default is 440@-10.

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Control Timer Values (sec)

Field

Description

Hook Flash Timer Min

Minimum on-hook time before off-hook qualifies as hook-flash. Less than this the

on-hook event is ignored. Range: 0.1–0.4 seconds.

The default is 0.1.

Hook Flash Timer Max

Callee On Hook Delay

Reorder Delay

Maximum on-hook time before off-hook qualifies as hook-flash. More than this the

on-hook event is treated as on-hook (no hook-flash event). Range: 0.4–1.6 seconds.

The default is 0.9.

Phone must be on-hook for at this time in sec before the SPA9000 will tear down the

current inbound call. It does not apply to outbound calls. Range: 0–255 seconds.

The default is 0.

Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf =

never plays. Range: 0–255 seconds.

The default is 5.

Call Back Expires

Call Back Retry Intvl

Call Back Delay

Expiration time in seconds of a call back activation. Range: 0–65535 seconds.

The default is 1800.

Call back retry interval in seconds. Range: 0–255 seconds.

The default is 30.

Delay after receiving the first SIP 18x response before declaring the remote end is

ringing. If a busy response is received during this time, the SPA9000 still considers the

call as failed and keeps on retrying.

The default is 0.5.

VMWI Refresh Intvl

Interdigit Long Timer

Interval between VMWI refresh to the CPE.

The default is 0.5.

Long timeout between entering digits when dialing. The interdigit timer values are used

as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all

valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.

The default is 10.

Interdigit Short Timer

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used

after any one digit, if at least one matching sequence is complete as dialed, but more

dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.

The default is 3.

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CPC Delay

Delay in seconds after caller hangs up when the SPA9000 starts removing the tip-and-ring

voltage to the attached equipment of the called party. Range: 0–255 seconds. SPA9000

has had polarity reversal feature since release 1.0 which can be applied to both the caller

and the callee end. This feature is generally used for answer supervision on the caller side

to signal to the attached equipment when the call has been connected (remote end has

answered) or disconnected (remote end has hung up). This feature should be disabled for

the called party (in other words, by using the same polarity for connected and idle state)

and the CPC feature should be used instead.

Without CPC enabled, reorder tone will is played after a configurable delay.

If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored

Resolution is 1 second.

The default is 2.

CPC Duration

Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up.

After that, tip-to-ring voltage is restored and dial tone applies if the attached equipment

is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second.

Resolution is 0.001 second.

The default is 0 (CPC disabled).

Vertical Service Activation Codes

Vertical Service Activation Codes are automatically appended to the dial-plan. There is no

need to include them in dial-plan, although no harm is done if they are included.

Field

Description

Call Return Code

This code calls the last caller.

The default is *69.

Call Redial Code

Redials the last number called. (Not in PAP2T)

The default is *07.

Blind Transfer Code

Begins a blind transfer of the current call to the extension specified after the activation

code.

The default is *98.

Call Back Act Code

Starts a callback when the last outbound call is not busy.

The default is *66.

Call Back Deact Code

Cancels a callback.

The default is *86.

Call Back Busy Act Code

Cfwd All Act Code

Starts a callback when the last outbound call is busy. (Not in PAP2T)

The default is *05

Forwards all calls to the extension specified after the activation code.

The default is *72.

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Cfwd All Deact Code

Cancels call forwarding of all calls.

The default is *73.

Cfwd Busy Act Code

Cfwd Busy Deact Code

Cfwd No Ans Act Code

Cfwd No Ans Deact Code

Cfwd Last Act Code

Forwards busy calls to the extension specified after the activation code.

The default is *90.

Cancels call forwarding of busy calls.

The default is *91.

Forwards no-answer calls to the extension specified after the activation code.

The default is *92.

Cancels call forwarding of no-answer calls.

The default is *93.

Forwards the last inbound or outbound calls to the extension specified after the activation

code.

The default is *63.

Cfwd Last Deact Code

Block Last Act Code

Block Last Deact Code

Accept Last Act Code

Cancels call forwarding of the last inbound or outbound calls.

The default is *83.

Blocks the last inbound call.

The default is *60.

Cancels blocking of the last inbound call.

The default is *80.

Accepts the last outbound call. It lets the call ring through when do not disturb or call

forwarding of all calls are enabled.

The default is *64.

Accept Last Deact Code

CW Act Code

Cancels the code to accept the last outbound call.

The default is *84.

Enables call waiting on all calls.

The default is *56.

CW Deact Code

Disables call waiting on all calls.

The default is *57.

CW Per Call Act Code

CW Per Call Deact Code

Block CID Act Code

Enables call waiting for the next call.

The default is *71.

Disables call waiting for the next call.

The default is *70.

Blocks caller ID on all outbound calls.

The default is *67.

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Block CID Deact Code

Removes caller ID blocking on all outbound calls.

The default is *68.

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.

The default is *81.

Block CID Per Call Deact Code Removes caller ID blocking on the next inbound call.

The default is *82.

Block ANC Act Code

Block ANC Deact Code

DND Act Code

Blocks all anonymous calls.

The default is *77.

Removes blocking of all anonymous calls.

The default is *87.

Enables the do not disturb feature.

The default is *78.

DND Deact Code

Disables the do not disturb feature.

The default is *79.

CID Act Code

Enables caller ID generation.

The default is *65.

CID Deact Code

Disables caller ID generation.

The default is *85.

CWCID Act Code

Enables call waiting, caller ID generation.

The default is *25.

CWCID Deact Code

Dist Ring Act Code

Dist Ring Deact Code

Speed Dial Act Code

Secure All Call Act Code

Secure No Call Act Code

Secure One Call Act Code

Disables call waiting, caller ID generation.

The default is *45.

Enables the distinctive ringing feature.

The default is *26

Disables the distinctive ringing feature.

The default is *46.

Assigns a speed dial number.

The default is *74.

Makes all outbound calls secure.

The default is *16.

Makes all outbound calls not secure.

The default is *17.

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by

default.)

The default is *18.

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Secure One Call Deact Code

Makes the next outbound call not secure. (It is redundant if all outbound calls are not

secure by default.)

The default is *19.

Conference Act Code

Attn-Xfer Act Code

If this code is specified, the user must enter it before dialing the third party for a

conference call. Enter the code for a conference call.

If the code is specified, the user must enter it before dialing the third party for a call

transfer. Enter the code for a call transfer.

Modem Line Toggle Code

Toggles the line to a modem.

The default is *99. Modem pass-through mode can be triggered only by pre-dialing this

code.

FAX Line Toggle Code

Referral Services Codes

Toggles the line to a fax machine. (Not in PAP2T)

The default is #99.

These codes tell the SPA9000 what to do when the user places the current call on hold and

is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123,

etc. Max total length is 79 chars. This parameter applies when the user places the current

call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the

following valid target number according to current dial plan) entered on the second

dial-tone triggers the SPA9000 to perform a blind transfer to a target number that is

prepended by the service *code.

For example, after the user dials *98, the SPA9000 plays a special dial tone called the

Prompt Tone while waiting for the user the enter a target number (which is checked

according to dial plan as in normal dialing). When a complete number is entered, the

SPA9000 sends a blind REFER to the holding party with the Refer-To target equals to

*98<target_number>. This feature allows the SPA9000 to hand off a call to an application

server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally

processed by the SPA9000. You can empty the corresponding *code that you do not want

to SPA9000 to process.

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Feature Dial Services Codes

These codes tell the SPA9000 what to do when the user is listening to the first or second

dial tone.

One or more *code can be configured into this parameter, such as *72, or

*72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user

has a dial tone (first or second dial tone). Enter *code (and the following target number

according to current dial plan) entered at the dial tone triggers the SPA9000 to call the

target number prepended by the *code. For example, after user dials *72, the SPA9000

plays a prompt tone awaiting the user to enter a valid target number. When a complete

number is entered, the SPA9000 sends a INVITE to *72<target_number> as in a normal

call. This feature allows the proxy to process features like call forward (*72) or BLock

Caller ID (*67).

The *codes should not conflict with any of the other vertical service codes internally

processed by the SPA9000. You can empty the corresponding *code that you do not want

to SPA9000 to process.

You can add a parameter to each *code in Features Dial Services Codes to indicate what

tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed

tone parameters (note the use of back quotes surrounding the parameter w/o spaces)

•

‘c‘ = <Cfwd Dial Tone>

‘d‘ = <Dial Tone>

‘m‘ = <MWI Dial Tone>

‘o‘ = <Outside Dial Tone>

‘p‘ = <Prompt Dial Tone>

‘s‘ = <Second Dial Tone>

‘x‘ = No tones are place, x is any digit not used above

If no tone parameter is specified, the SPA9000 plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call

forwarding, do not include it in this parameter. In that case, simple add that *code in the

dial plan and the SPA9000 send INVITE *73@..... as usual when user dials *73.

Vertical Service Announcement Codes

Field

Description

Service Annc Base Number

Base number for service announcements.

Service Annc Extension Codes Extension codes for service announcements.

Outbound Call Codec Selection Codes

These codes automatically appended to the dial-plan. So no need to include them in dial-plan (although

no harm to do so either).

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Field

Description

Prefer G711u Code

Force G711u Code

Prefer G711a Code

Force G711a Code

Prefer G723 Code

Makes this codec the preferred codec for the associated call.

The default is *017110.

Makes this codec the only codec that can be used for the associated call.

The default is *027110.

Makes this codec the preferred codec for the associated call.

The default is *017111

Makes this codec the only codec that can be used for the associated call.

The default is *027111.

Makes this codec the preferred codec for the associated call.

The default is *01723.

Force G723 Code

Makes this codec the only codec that can be used for the associated call.

The default is *02723.

Prefer G726r16 Code

Force G726r16 Code

Prefer G726r24 Code

Force G726r24 Code

Prefer G726r32 Code

Force G726r32 Code

Prefer G726r40 Code

Force G726r40 Code

Prefer G729a Code

Force G729a Code

Makes this codec the preferred codec for the associated call.

The default is *0172616.

Makes this codec the only codec that can be used for the associated call.

The default is *0272616.

Makes this codec the preferred codec for the associated call.

The default is *0172624.

Makes this codec the only codec that can be used for the associated call.

The default is *0272624.

Makes this codec the preferred codec for the associated call.

The default is *0172632.

Makes this codec the only codec that can be used for the associated call.

The default is *0272632.

Makes this codec the preferred codec for the associated call.

The default is *0172640.

Makes this codec the only codec that can be used for the associated call.

The default is *0272640.

Makes this codec the preferred codec for the associated call.

The default is *01729.

Makes this codec the only codec that can be used for the associated call.

The default is *02729.

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Miscellaneous

Field

Description

Set Local Date (mm/dd)

Sets 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)

Time Zone

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

Selects the number of hours to add to GMT to generate the local time for caller ID

generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00,

…, GMT+13:00.

The default is GMT-08:00.

FXS Port Impedance

Sets the electrical impedance of the FXS port. Choices are 600, 900, 600+2.16uF,

900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 200+600||100nF.

The default is 600.

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.

Daylight Saving Time Rule

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]]]

•

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.

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FXS Port Input Gain

Input gain in dB, up to three decimal places. The range is 6.000 to -12.000.

The default is -3.

FXS Port Output Gain

Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call

Progress Tones and DTMF playback level are not affected by the <FXS Port Output

Gain>.

The default is -3.

DTMF Playback Level

DTMF Playback Length

Detect ABCD

Local DTMF playback level in dBm, up to one decimal place.

The default is -16.0.

Local DTMF playback duration in milliseconds.

The default is .1.

To enable local detection of DTMF ABCD, select yes. Otherwise, select no.

The default is yes. Setting has no effect if DTMF Tx Method is INFO; ABCD is always

sent OOB regardless in this setting.

Playback ABCD

Caller ID Method

To enable local playback of OOB DTMF ABCD, select yes. Otherwise, select no.

The default is yes.

The following choices are available:

•

Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent after first ring (same

as ETSI FSK sent after first ring) (no polarity reversal or DTAS).

DTMF (Finland, Sweden)—CID only. DTMF sent after polarity reversal (and no

DTAS) and before first ring.

DTMF (Denmark)—CID only. DTMF sent after polarity reversal (and no DTAS) and

before first ring.

ETSI DTMF—CID only. DTMF sent after DTAS (and no polarity reversal) and

before first ring.

ETSI DTMF With PR—CID only. DTMF sent after polarity reversal and DTAS and

before first ring.

ETSI DTMF After Ring—CID only. DTMF sent after first ring (no polarity reversal

or DTAS).

ETSI FSK—CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity

reversal) and before first ring. Waits for ACK from CPE after DTAS for CIDCW.

ETSI FSK With PR (UK)—CID, CIDCW, and VMWI. FSK is sent after polarity

reversal and DTAS and before first ring. Waits for ACK from CPE after DTAS for

CIDCW. Polarity reversal is applied only if equipment is on hook.

The default is Bellcore(N.Amer, China).

Caller ID FSK Standard

The SPA9000 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. (Not in PAP2T)

The default is Default.

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More Echo Suppression

Enable or disable more echo suppresion. The default is no.

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FXS 1/2 Tab

This section describes the fields for the following headings on the FXS 1 and FXS 2 tabs:

•

Line Enable, page 6-38

Network Settings, page 6-38

SIP Settings, page 6-39

Subscriber Information, page 6-41

Dial Plan, page 6-41

Mailbox Status, page 6-41

Streaming Audio Server (SAS), page 6-42

Call Feature Settings, page 6-43

Audio Configuration, page 6-43

FXS Port Polarity Configuration, page 6-46

Line Enable

Field

Description

Line Enable

To enable this line for service, select yes. Otherwise, select no.

The default is yes.

Network Settings

Field

Description

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.

The default is 0x68.

SIP CoS Value [0-7]

RTP ToS/DiffServ Value

RTP CoS Value [0-7]

CoS value for SIP messages.

The default is 3.

ToS/DiffServ field value in UDP IP packets carrying RTP data.

The default is 0xb8.

CoS value for RTP data.

The default is 6.

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FXS 1/2 Tab

Network Jitter Level

Determines how jitter buffer size is adjusted by the SPA9000. 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

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

Field

Description

SIP Port

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.

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SIP Debug Option

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

which SIP messages to log. The choices are as follows:

•

none—No logging.

1-line—Logs the start-line only for all messages.

1-line excl. OPT—Logs the start-line only for all messages except OPTIONS

requests/responses.

•

1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY

requests/responses.

1-line excl. REG—Logs the start-line only for all messages except REGISTER

requests/responses.

1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except

OPTIONS, NOTIFY, and REGISTER

requests/responses.

•

full—Logs all SIP messages in full text.

full excl. OPT—Logs all SIP messages in full text except OPTIONS

requests/responses.

•

full excl. NTFY—Logs all SIP messages in full text except NOTIFY

requests/responses.

full excl. REG—Logs all SIP messages in full text except REGISTER

requests/responses.

full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS,

NOTIFY, and REGISTER requests/responses.

The default is none.

RTP Log Intvl

Periodically, the SPA9000 logs RTP statistics via syslog, depending on debug level. Enter

the period of time in seconds.

The default is 0.

Restrict Source IP

If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled,

the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines.

To enable the Restrict Source IP feature, select yes. Otherwise, select no.

The default is no.

Referor Bye Delay

Controls when the SPA9000 sends BYE to terminate stale call legs after completion of

call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To

Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate

period of time in seconds.

The default is 4.

Refer Target Bye Delay

Referee Bye Delay

For the Refer Target Bye Delay, enter the appropriate period of time in seconds.

The default is 0.

For the Referee Bye Delay, enter the appropriate period of time in seconds.

The default is 0.

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Refer-To Target Contact

Sticky 183

To contact the refer-to target, select yes. Otherwise, select no.

The default is no.

If this feature is enabled, the IP telephony ignores further 180 SIP responses after

receiving the first 183 SIP response for an outbound INVITE. To enable this feature,

select yes. Otherwise, select no.

The default is no.

Subscriber Information

Field

Description

Display Name

User ID

Display name for caller ID.

Extension number for this line.

Dial Plan

Field

Description

Dial Plan

Dial plan script for this line.

Mailbox Status

Field

Description

Message Waiting

Indicates whether you have new voicemail waiting.

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Streaming Audio Server (SAS)

Field

Description

SAS Enable

To enable the use of the line as a streaming audio source, select yes. Otherwise, select no.

If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming

calls and streams audio RTP packets to the caller.

The default is no.

SAS DLG Refresh Intvl

SAS Inbound RTP Sink

If this is not zero, it is the interval at which the streaming audio server sends out session

refresh (SIP re-INVITE) messages to determine whether the connection to the caller is

still active. If the caller does not respond to the refresh message, the SPA9000 ends this

call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session

refresh is disabled).

The default is 30.

This setting works around devices that do not play inbound RTP if the streaming audio

server line declares itself as a send-only device and tells the client not to stream out audio.

Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this is used

by the SPA9000’s streaming audio server line in the SDP of its 200 response to an inbound

INVITE message from a client.

The purpose of this parameter is to work around devices that do not play inbound RTP if

the SAS line declares itself as a “sendonly” device and tells the client not to stream out

audio. This parameter is a FQDN or IP address of a RTP sink to be used by the SPA SAS

line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the

c = line and the port number and, if specified, in the m = line of the SDP. If this value is

not specified or equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell

the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified,

then a=sendrecv and the SAS client will stream audio to the given address. Special case:

If the value is $IP, then the SAS line’s own IP address is used in the c = line and

a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line.

The default value is empty.

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Call Feature Settings

Field

Description

Blind Attn-Xfer Enable

Enables the SPA9000 to perform an attended transfer operation by ending the current call

leg and performing a blind transfer of the other call leg. If this feature is disabled, the

SPA9000 performs an attended transfer operation by referring the other call leg to the

current call leg while maintaining both call legs. To use this feature, select yes. Otherwise,

select no.

The default is no.

MOH Server

User ID or URL of the auto-answering streaming audio server. When only a user ID is

specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the

MOH Server is not specified.

Xfer When Hangup Conf

Makes the SPA9000 perform a transfer when a conference call has ended. Select yes or

no from the drop-down menu.

The default is Yes.

Conference Bridge URL

This feature supports external conference bridging for n-way conference calls (n > 2),

instead of mixing audio locally. To use this feature, set this parameter to that of the

server's name; for example, conf@myserver.com:12345 or conf (which uses the Proxy

value as the domain).

Conference Bridge Ports

Enable IP Dialing

Maximum number of conference call participants. The range is 3 to 10.

The default is 3.

To use IP dialing, select yes. Otherwise, select no.

The default is no.

Emergency Number

Comma-separated list of emergency number patterns. If the outbound call matches one of

the patterns, the SPA9000 disables hook flash event handling. Hook flash event handling

is restored to normal when the phone is on-hook again. If you leave this field blank, the

SPA9000 has no emergency number.

Mailbox ID

Enter the ID number of the mailbox for this line.

Audio Configuration

A codec resource is considered as allocated if it has been included in the SDP codec list of an active call,

even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is

enabled and included in the codec list, that resource is tied up until the end of the call whether or not the

call actually uses G.729a. If the G729a resource is already allocated and since only one G.729a resource

is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices

are G711a and G711u. On the other hand, two G.723.1/G.726 resources are available per device.

Therefore it is important to disable the use of G.729a in order to guarantee the support of two

simultaneous G.723/G.726 codec.

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Field

Description

Preferred Codec

Preferred codec for all calls. (The actual codec used in a call still depends on the outcome

of the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-16,

G726-24, G726-32, G726-40, G729a, or G723.

The default is G711u.

Silence Supp Enable

Use Pref Codec Only

To enable silence suppression so that silent audio frames are not transmitted, select yes.

Otherwise, select no.

The default is no.

To use only the preferred codec for all calls, select yes. (The call fails if the far end does

not support this codec.) Otherwise, select no.

The default is no.

Silence Threshold

G729a Enable

Select the appropriate setting for the threshold: high, medium, or low.

The default is medium.

To enable the use of the G729a codec at 8 kbps, select yes. Otherwise, select no.

The default is yes.

Echo Canc Enable

G723 Enable

To enable the use of the echo canceller, select yes. Otherwise, select no.

The default is yes.

To enable the use of the G723a codec at 6.3 kbps, select yes. Otherwise, select no.

The default is yes.

Echo Canc Adapt Enable

G726-16 Enable

To enable the echo canceller to adapt, select yes. Otherwise, select no.

The default is yes.

To enable the use of the G726 codec at 16 kbps, select yes. Otherwise, select no.

The default is yes.

Echo Supp Enable

G726-24 Enable

To enable the use of the echo suppressor, select yes. Otherwise, select no.

The default is yes.

To enable the use of the G726 codec at 24 kbps, select yes. Otherwise, select no.

The default is yes.

FAX CED Detect Enable

To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise,

select no.

The default is yes.

G726-32 Enable

To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no.

The default is yes.

FAX CNG Detect Enable

G726-40 Enable

To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.

The default is yes.

To enable the use of the G726 codec at 40 kbps, select yes. Otherwise, select no.

The default is yes.

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FAX Passthru Codec

DTMF Process INFO

FAX Codec Symmetric

Select the codec for fax passthrough, G711u or G711a.

The default is G711u.

(Not in PAP2T) To use the DTMF process info feature, select yes. Otherwise, select no.

The default is yes.

To force the SPA9000 to use a symmetric codec during fax passthrough, select yes.

Otherwise, select no.

The default is yes.

DTMF Process AVT

FAX Passthru Method

DTMF Tx Method

(Not in PAP2T) To use the DTMF process AVT feature, select yes. Otherwise, select no.

The default is yes.

Select the fax passthrough method: None, NSE, or ReINVITE.

The default is NSE.

Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto,

InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends

DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based

on the outcome of codec negotiation.

The default is Auto.

FAX Process NSE

To use the fax process NSE feature, select yes. Otherwise, select no.

The default is yes.

Hook Flash Tx Method

Select the method for signaling hook flash events: None, AVT, or INFO. None does not

signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO

with the single line signal=hf in the message body. The MIME type for this message body

is taken from the Hook Flash MIME Type setting.

The default is None.

FAX Disable ECAN

If enabled, this feature automatically disables the echo canceller when a fax tone is

detected. To use this feature, select yes. Otherwise, select no.

The default is no.

Release Unused Codec

This feature allows the release of codecs not used after codec negotiation on the first call,

so that other codecs can be used for the second line. To use this feature, select yes.

Otherwise, select no.

The default is yes.

FAX Enable T38

To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no.

The default is yes.

FAX Tone Detect Mode

This parameter has three possible values:

•

caller or callee - SPA will detect FAX tone whether it is callee or caller

caller only - SPA will detect FAX tone only if it is the caller

callee only - SPA will detect FAX tone only if it is the callee

The default is caller or callee.

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FXS Port Polarity Configuration

Field

Description

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

The default is Forward.

Caller Conn Polarity

Callee Conn Polarity

Polarity after an outbound call is connected: Forward or Reverse.

The default is Forward.

Polarity after an inbound call is connected: Forward or Reverse.

The default is Forward.

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This section describes the fields for the following headings on the Line 1, Line 2, Line 3, and Line 4

tabs:

•

Line Enable, page 6-47

Network Settings, page 6-47

SIP Settings, page 6-48

Subscriber Information, page 6-50

Dial Plan, page 6-52

Note

In a configuration profile, the Line parameters must be appended with [1] or [2] to identify the line to

which the setting applies.

Line Enable

Field

Description

Line Enable

To enable this line for service, select yes. Otherwise, select no.

The default is yes.

Network Settings

Field

Description

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.

The default is 0x68.

SIP CoS Value [0-7]

RTP ToS/DiffServ Value

RTP CoS Value [0-7]

CoS value for SIP messages.

The default is 3.

ToS/DiffServ field value in UDP IP packets carrying RTP data.

The default is 0xb8.

CoS value for RTP data.

The default is 6.

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SIP Settings

Field

Description

SIP Port

Port number of the SIP message listening and transmission port.

The default is 5060.

SIP 100REL Enable

To enable the support of 100REL SIP extension for reliable transmission of provisional

responses (18x) and use of PRACK requests, select yes. Otherwise, select no.

The default is no.

EXT SIP Port

The external SIP port number.

Auth Resync-Reboot

If this feature is enabled, the SPA9000 authenticates the sender when it receives the

NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise,

select no.

The default is yes.

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it sees this header from

the user agent. If this field is configured and the proxy does not support it, it responds

with the message, unsupported. Enter the appropriate header in the field provided.

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 GUID

The Global Unique ID is generated for each line for each device. When it is enabled, the

SPA9000 adds a GUID header in the SIP request. The GUID is generated the first time

the unit boots up and stays with the unit through rebooting and even factory reset. This

feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts.

The default is yes.

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SIP Debug Option

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

which SIP messages to log. Choices are as follows:

•

none—No logging.

1-line—Logs the start-line only for all messages.

1-line excl. OPT—Logs the start-line only for all messages except OPTIONS

requests/responses.

•

1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY

requests/responses.

1-line excl. REG—Logs the start-line only for all messages except REGISTER

requests/responses.

1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except

OPTIONS, NOTIFY, and REGISTER

requests/responses.

•

full—Logs all SIP messages in full text.

full excl. OPT—Logs all SIP messages in full text except OPTIONS

requests/responses.

•

full excl. NTFY—Logs all SIP messages in full text except NOTIFY

requests/responses.

full excl. REG—Logs all SIP messages in full text except REGISTER

requests/responses.

full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS,

NOTIFY, and REGISTER requests/responses.

The default is none.

RTP Log Intvl

The interval for the RTP log.

Restrict Source IP

If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled,

the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines.

To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured,

the PAP2T will drop all packets sent to its SIP Ports originated from an

untrusted IP address. A source IP address is untrusted if it does not match any

of the IP addresses resolved from the configured <Proxy> (or <Outbound

Proxy> if <Use Outbound Proxy> is yes).

The default is no.

Referor Bye Delay

Controls when the SPA9000 sends BYE to terminate stale call legs upon completion of

call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To

Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate

period of time in seconds.

The default is 4.

Refer Target Bye Delay

Referee Bye Delay

For the Refer Target Bye Delay, enter the appropriate period of time in seconds.

The default is 0.

For the Referee Bye Delay, enter the appropriate period of time in seconds.

The default is 0.

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Refer-To Target Contact

To contact the refer-to target, select yes. Otherwise, select no.

The default is no.

Subscriber Information

Field

Description

Display Name

User ID

Display name for caller ID.

Extension number for this line.

Password for this line.

Password

Use Auth ID

To use the authentication ID and password for SIP authentication, select yes. Otherwise,

select no to use the user ID and password.

The default is no.

Auth ID

Authentication ID for SIP authentication.

Call Capacity

Maximum number of calls allowed on this line interface. Choices: {unlimited,1,2,3,…25

}. Default is 16. Note that the SPA9000 does not distinguish between incoming and

outgoing calls when talking about call capacity. Note: unlimited = 16

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Contact List

List of client stations that the SPA9000 should alert when there is an incoming call to the

line interface from the ITSP. Each rule is also known as a hunt group and the did is the

group’s extension.

The default method to ring a group is to ring all the members simultaneously, unless a

hunt rule is specified. The format is rule[|rule[|rule[...]]] where:

•

rule = [did:]ext[,ext[,ext[…]]][,name=gname][,hunt=hrule][,cfwd=target]

did = Embedded DID number. If did is not specified, the rule applies to any DID

number

•

ext = Client extension number pattern; may contain * or ? wildcard or %xx escaped

characters

If an hrule is specified, the listed stations shall be contacted sequentially (known as

hunting); otherwise all the listed stations ring simultaneously. A hrule has the

following format: hrule=algo;interval;max, where

–

•

algo determines the order to ring the stations. It can be one of the following:

restart or re—Always start from beginning of the list,

•

next or ne—Start from the next on the list to the last station that rings,

random or ra—Order is randomized for each call

•

–

interval is the time to ring each station in seconds

max is the total time in seconds to hunt before rejecting the call or forwarding it

to voice mail. If max is less than interval, it is interpreted as the number of cycles

to go through the hunt group before hunting stops. If max is 0, hunting goes on

indefinitely until the caller hangs up or some one answers the call

The list is recycled until the total hunt time or number of cycles is reached, whichever

comes first. For example: hunt=next;12;1, where:

•

target—A user-id to forward the caller if no one replies or the client stations has rung

for a delay equal to the value specified in <Cfwd No Ans Delay>. If target starts with

vm, such as vm23456, the call is forwarded to voicemail with the mailbox ID equal

to the rest of the digits (note that the first digit indicates the line interface to use to

access this mailbox). If ,cfwd=… does not present, the call fails without being

forwarded and <Cfwd No Ans Delay> does not apply.

•

gname—A name to associate with the hunt group. It can be a token such as Sales or

a quoted string, such as The Sales Team.

Note that each member of a rule can be the extension of a group defined in <Hunt Group>.

Examples:

•

530?,hunt=ra;10;1,cfwd=vm5404

5300, cfwd=vm5300

4089993001:5001|4089993000:5000,cfwd=aa

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Contact List (cont.)

Default is blank.

Since version 3.2.8, SPA9000 supports Phone Extension to DID Number Mapping

through an added syntax in this parameter. If such mapping is provided for a given

extension, the user-id and display name fields in the FROM and REFERRED-BY headers

(wherever applied) of outbound requests originated from that phone to the ITSP are

replaced by the corresponding DID number and the phone’s assigned display name

respectively. If the mapping is not provided, the SPA9000 uses the line interface’s user-id

and display name instead (old behavior). To specify the mapping from Extension to DID,

insert a ‘+’ or ‘-’ in front of the ext of a rule. The ext must not include any wildcard

characters. If ‘+’ is used, the ext is alerted when the corresponding DID is called (just like

the case without the ‘+’ prefix). If ‘-’ is used, the ext is not be alerted (that is, it is only

there to define a ext-to-DID-mapping). Note that ‘+’ or ‘-’ may be assigned to more than

one ext in the same rule. For example:

4089991002:1001,+1002|4089991008:aa,-1008

For this example, when 1002 makes an external call, the FROM user-id shows

4089991002. When 1008 makes an external call, the FROM user-id shows 4089991008.

When 4089991002 is called, both 1001 and 1002 are alerted. When 4089991008 is called,

only the AA is alerted.

Cfwd No Ans Delay

Delay, in seconds, before the call forwarding of no-answer calls feature is triggered.

The default is 20.

Dial Plan

The default dial plan script for each line is as follows:

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.). The syntax for a dial plan

expression is as follows:

Table 6-1 Dial Plan Script Syntax

Dial Plan Entry

*xx

Functionality

Allow arbitrary 2 digit star code

Allow x11 sequences

[3469]11

Operator

Int’l Operator

[2-9]xxxxxx

1xxx[2-9]xxxxxx

xxxxxxxxxxxx.

US local number

US 1 + 10-digit long distance number

Everything else (Int’l long distance, FWD, ...)

If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by

entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and

port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the

user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number

according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is

enabled.

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Field

Description

Dial Plan

Dial plan script for this line.

The default is (<9:>xx.)

The dial plan syntax is expanded in the SPA3102 to allow the designation of three

parameters to be used with a specific gateway:

•

uid – the authentication user-id

pwd – the authentication password

nat – if this parameter is present, use NAT mapping

Each parameter is separated by a semi-colon (;).

Furthermore, it recognizes gw0, gw1, …, gw4 as the locally configured gateways, where

gw0 represents the local PSTN gateway in the same SPA3102 unit.

Example 1:

*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xyz

Example 2:

*1xxxxxxxxxx<:@fwd.pulver.com;nat;uid=jsmith;pwd=xyz

Example 3:

[39]11<:@gw0>

NAT Settings

Field

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.

Otherwise, select no.

The default is no.

NAT Keep Alive Enable

NAT Keep Alive Msg

To send the configured NAT keep alive message periodically, select yes. Otherwise, select

no.

The default is no.

Enter the keep alive message that should be sent periodically to maintain the current

NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is

$REGISTER, a REGISTER message without contact is sent.

The default is $NOTIFY.

Destination that should receive NAT keep alive messages. If the value is $PROXY,

the messages are sent to the current or outbound proxy.

NAT Keep Alive Dest

The default is $PROXY.

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Chapter 6 SPA9000 Field Reference

Line 1/2/3/4 Tab

Enter the port number of the external port to substitute for the actual SIP port of the

EXT SIP Port

SPA9000 in all outgoing SIP messages.

The default is 0.

Proxy and Registration

Field

Description

Proxy

SIP proxy server for all outbound requests.

Use Outbound Proxy

Enable the use of <Outbound Proxy>. If set to no, <Outbound Proxy> and <Use OB Proxy

in Dialog) is ignored.

The default is no.

Outbound Proxy

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.

Use OB Proxy In Dialog

Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored

if <Use Outbound Proxy> is no or <Outbound Proxy> is empty.

The default is yes.

Enable periodic registration with the <Proxy>. This parameter is ignored if <Proxy> is not

specified.

The default is yes.

Make Call Without Reg

Allow making outbound calls without successful (dynamic) registration by the unit. If No,

dial tone will not play unless registration is successful.

The default is no.

Allow answering inbound calls without successful (dynamic) registration by the unit. If

proxy responded to REGISTER with a smaller Expires value, the PAP2T will renew

registration based on this smaller value instead of the configured value. If registration

failed with an Expires too brief error response, the PAP2T will retry with the value given

in the Min-Expires header in the error response.

The default is 30.

Ans Call Without Reg

Expires value in sec in a REGISTER request. PAP2T will periodically renew registration

shortly before the current registration expired. This parameter is ignored if <Register> is

no. Range: 0 – (231 – 1) sec

Use DNS SRV

Whether to use DNS SRV lookup for Proxy and Outbound Proxy.

The default is no.

DNS SRV Auto Prefix

If enabled, the PAP2T will automatically prepend the Proxy or Outbound Proxy name with

_sip._udp when performing a DNS SRV lookup on that name.

The default is no.

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Field

Description

Proxy Fallback Intvl

This parameter sets the delay (sec) after which the PAP2T will retry from the highest

priority proxy (or outbound proxy) servers after it has failed over to a lower priority

server. This parameter is useful only if the primary and backup proxy server list is

provided to the PAP2T via DNS SRV record lookup on the server name. (Using multiple

DNS A record per server name does not allow the notion of priority and so all hosts will

be considered at the same priority and the PAP2T will not attempt to fall back after a fail

over).

The default is 3600

Proxy Redundancy Method

PAP2T will make an internal list of proxies returned in DNS SRV records. In normal mode

this list will contain proxies hierarchied by weight and priority

if Based on SRV port is configured the PAP2T does normal first, and also inspect the port

number based on 1^stproxy’s port on the list.

The default is Normal.

Voice Mail Server

Identifies the voicemail server.

Mailbox Subscribe Expires

Identifies when the mailbox subscription expires.

The default is 30.

Mailbox Subscribe URL

Mailbox Deposit URL

Mailbox Manage URL

Mailbox Status

URL that should receive the SUBSCRIBE messages, so the SPA9000 receives voicemail

status notification for all mailboxes on this line.

Note: This parameter allows $USER and $PROXY macros, such as $USER@$PROXY

URL that the SPA9000 contacts when clients and external callers need to deposit

voicemail in any of the mailboxes on this line.

Default is blank. This parameter allows $USER, $PROXY, and $MBID macros, such as

$USER@$PROXY, or $MBID@$PROXY.

URL that the IP telephony contacts when it needs to check voicemail for any of the

mailboxes on this line.

Default is blank. This parameter allows $USER, $PROXY, and $MBID macros, such as

$USER@$PROXY, or $MBID@$PROXY.

Mailbox status for all the mailboxes associated with this line interface. The status is

automatically updated when the SPA9000 receives voicemail status notification from the

ITSP.

Format: [mbs[,mbs[,mbs[,…]]]] where:

•

mbs = mbid:new/old

mbid = mailbox ID, such as 12345

new = number of new messages in mailbox: 0,1,2,…

old = number of old messages in mailbox: 0, 1, 2, …

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Field

Description

VMSP Bridge

Applies only if this line interface offers voice mail services. It specifies whether the

SPA9000 should bridge the call from an external caller on the same or a different line

interface. The choices are:

•

None—Do not bridge external calls from any line interfaces

All—Bridge external calls from any line interfaces except from the same line

All+Self—Bridge external calls from any line interfaces including the same line

Notes:

•

If the external call is also on this line interface, the SPA9000 does not attempt to

bridge the call even if the value is All.

•

If <PBX Network Interface> is WAN, <Force Media Proxy> must be yes for VMSP

Bridging to function properly.

The default is None.

CFWD Bridge Mode

Instructs the SPA9000 how to handle call forwarding of an external caller to another

external number by a client station. This parameter was added in Release 5.1.

The normal way of performing this operation is for the SPA9000 to send a (blind) SIP

REFER to the calling device to let it contact the target number directly. It then drops out

of the call completely. This requires the calling device to understand the SIP signaling

involved and the operation permitted by the underlying service provider. The SPA400, for

instance, cannot handle this operation.

With bridging, the SPA9000 maintains two separate call legs throughout the call: one with

the caller and one with the call forward target. The two call peers connect only with the

SPA9000, while the SPA9000 acts as a proxy for the RTP packets exchanged between the

two parties.

This parameter has two possible values:

•

None—Do not bridge forwarded calls (use the normal REFER method)

All—Bridge all forwarded calls

The default is None.

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Field

Description

XFER Bridge Mode

Instructs the SPA9000 how to handle call transferring of an external caller to another

external number by a client station. This parameter was added in Release 5.1.

The normal way of performing this operation is for the SPA9000 to send a SIP REFER

method to the calling device to let it contact the transfer target directly. The SPA9000 then

drops out of the call completely. This requires the calling device (the transferee) and the

target device to understand the SIP signaling involved and the operation permitted by the

underlying service providers. Note that the call legs with transferee and the transfer target

might be with different ITSP. The SPA400, for instance, cannot handle this operation.

With bridging, the SPA9000 maintains two separate call legs throughout the call: one with

the transferred call and one with the transfer target. The two call peers connect only with

the SPA9000, while the SPA9000 acts as a proxy for the RTP packets exchanged between

the two parties.

This parameter has three possible values:

•

none —Do not bridge call transfer (use the normal REFER method)

all —Bridge all call transfer

all except same line—Bridge call transfer only if it is between 2 different Line

interfaces

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A P P E N D I X

Acronyms

A/D

Analog To Digital Converter

ANC

B2BUA

Bool

Anonymous Call

Back to Back User Agent

Boolean Values. Specified as “yes” and “no”, or “1” and “0” in the profile

Certificate Authority

CAS

CPE Alert Signal

CDR

CID

Call Detail Record

Caller ID

CIDCW

CNG

CPC

Call Waiting Caller ID

Comfort Noise Generation

Calling Party Control

CPE

Customer Premises Equipment

Call Waiting Caller ID

CWCID

CWT

D/A

Call Waiting Tone

Digital to Analog Converter

decibel

dBm

dB with respect to 1 milliwatt

Dynamic Host Configuration Protocol

Domain Name Server

DHCP

DNS

DRAM

DSL

Dynamic Random Access Memory

Digital Subscriber Loop

DSP

Digital Signal Processor

DTAS

DTMF

FQDN

FSK

Data Terminal Alert Signal (same as CAS)

Dual Tone Multiple Frequency

Fully Qualified Domain Name

Frequency Shift Keying

FXS

Foreign eXchange Station

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Appendix A Acronyms

Gateway

ITU

International Telecommunication Union

Hypertext Markup Language

Hypertext Transfer Protocol

HTTP over SSL

HTML

HTTP

HTTPS

ICMP

IGMP

ILEC

Internet Control Message Protocol

Internet Group Management Protocol

Incumbent Local Exchange Carrier

Internet Protocol

ISP

Internet Service Provider

IP Telephony Service Provider

Interactive Voice Response

Local Area Network

ITSP

IVR

LAN

LBR

LBRC

Low Bit Rate

Low Bit Rate Codec

Mini-Certificate

MGCP

MOH

MOS

Media Gateway Control Protocol

Music On Hold

Mean Opinion Score (1-5, the higher the better)

Millisecond

MSA

MWI

OSI

Music Source Adaptor

Message Waiting Indication

Open Switching Interval

Printed Circuit Board

PCB

Polarity Reversal

Provisioning Server

PSQM

PSTN

NAT

OOB

REQT

RESP

RSC

RTP

Perceptual Speech Quality Measurement (1-5, the lower the better)

Public Switched Telephone Network

Network Address Translation

Out-of-band

(SIP) Request Message

(SIP) Response Message

(SIP) Response Status Code, such as 404, 302, 600

Real Time Protocol

RTT

Round Trip Time

SAS

Streaming Audio Server

SDP

Session Description Protocol

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Appendix A Acronyms

SDRAM

sec

Synchronous DRAM

seconds

SIP

Session Initiation Protocol

Shared line appearance

Subscriber Line Interface Circuit

Service Provider

SLA

SLIC

SPA

SSL

Linksys Phone Adaptor

Secure Socket Layer

Trivial File Transfer Protocol

Transmission Control Protocol

User Agent

TFTP

TCP

Micro-controller

UDP

URL

User Datagram Protocol

Uniform Resource Locator

Voicemail

VMWI

Visual Message Waiting Indication/Indicator

Voice Quality

WAN

XML

Wide Area Network

Extensible Markup Language

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A P P E N D I X

Glossary

ACD (Automatic Call Distribution)—A switching system designed to allocate incoming calls to certain

positions or agents in the order received and to hold calls not ready to be handled (often with a recorded

announcement).

Area code—A 3-digit code used in North America to identify a specific geographic telephone location.

The first digit can be any number between 2 and 9. The second and third digits can be any number.

Billing increment—The division by which the call is rounded. In the field it is common to see

full-minute billing on the local invoice while 6-second rounding is the choice of most long-distance

providers that bill their customers directly.

Blocked calls—Caused by an insufficient network facility that does not have enough lines to allow calls

to reach a given destination. May also pertain to a call from an originating number that is blocked by the

receiving telephone number.

Bundled service—Offering various services as a complete package.

Call completion—The point at which a dialed number is answered.

Call termination—The point at which a call is disconnected.

CDR (Call Detail Records)—A software program attached to a VoIP/telephone system that records

information about the telephone number’s activity.

Carrier’s carrier—Companies that build fiber optic and microwave networks primarily selling to

resellers and carriers. Their main focus is on the wholesale and not the retail market.

Casual access—When customers choose not to use their primary carriers to process the long-distance

call being made. The customer dials the carrier’s 101XXXX number.

CO (Central Office)—Switching center for the local exchange carrier.

Centrex—This service is offered by the LEC to the end user. The feature-rich Centrex line offers the

same features and benefits as a PBX to a customer without the capital investment or maintenance

charges. The LEC charges a monthly fee to the customer, who must agree to sign a term agreement.

Circuits—The communication path(s) that carry calls between two points on a network.

Customer Premise Equipment—The only part of the telecommunications system that the customer

comes into direct contact with. Example of such pieces of equipment are telephones, key systems, PBXs,

voice-mail systems, and call accounting systems as well as wiring telephone jacks. The standard for this

equipment is set by the FCC, and the equipment is supplied by an interconnect company.

Dedicated access—Customers have direct access to the long-distance provider via a special circuit (T1

or private lines). The circuit is hardwired from the customer site to the POP and does not pass through

the LEC switch. The dial tone is provided from the long-distance carrier.

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Appendix B Glossary

Dedicated Access Line (DAL)—Provided by the local exchange carrier. An access line from the

customer’s telephone equipment directly to the long-distance company’s switch or POP.

Demarcation point—This is where the LEC ownership and responsibility (wiring, equipment) ends and

the customer’s responsibilities begin.

Direct Inward Dialing (DID)—Allows an incoming call to bypass the attendant and ring directly to an

extension. Available on most PBX systems and a feature of Centrex service.

Dual Tone Multifrequency (DTMF)—Better known as the push button keypad. DTMF replaces dial

pulses with electronically produced tones for network signaling.

Enhanced service—Services that are provided in addition to basic long distance and accessed by way of

a touchtone phone through a series of menus.

Exchange code (NXX)—The first three digits of a phone number.

Flat-rate pricing—The customer is charged one rate (sometimes two rates, one for peak and one for

off-peak) rather than a mileage-sensitive program rate.

IXC (Interexchange Carrier)—A long-distance provider that maintains its own switching equipment.

IVR (Interactive Voice Response)—Provides a mechanism for information to be stored and retrieved

using voice and a touchtone telephone.

Local loop—The local telephone company provides the transmission facility from the customer to the

telephone company’s office, which is engineered to carry voice and/or data.

North American Numbering Plan (NANP)—How telephone numbers are identified in North America.

The telephone number can be identified based on their three separate components: (NPA), (NXX), and

(XXXX).

PIN (Personal Identification Code)—A customer calling/billing code for prepaid and pay-as-you-go

calling cards.

Private Branch Exchange—Advanced phone system commonly used by the medium to larger customer.

It allows the customer to perform a variety of in-house routing (inside calling). The dial tone that is heard

when the customer picks up the phone is an internal dial tone.

SS7 (SPA9000 Signaling Number 7)—Technology used by large carriers to increase the reliability and

speed of transmission between switches.

Switch (switching)—Equipment that connects and routes calls and provides other interim functions such

as least cost routing, IVR, and voicemail. It performs the “traffic cop” function of telecommunications

via automated management decisions.

Touchtone (DTMF)—The tone recognized by a push button (touchtone) telephone.

Unified messaging— Platform that lets users send, receive, and manage all e-mail, voice, and fax

messages from any telephone, PC, or information device.

Voicemail—A system that allows storage and retrieval of voice messages through voice-mail boxes.

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I N D E X

Setup Wizard 2-15

Admin Passwd 6-6

advanced features, configuring 2-18

analog telephone 2-4, 2-16

architecture, SPA9000 1-12

ATA 1-13

Nu m e ric s

16-user license 2-13

300 response 4-14

404 Forbidden 2-33

73738 resetting factory defaults 2-8

7932 enabling the administration web server 2-16

Attn-Xfer Act Code 6-32

audio 5-23

audio file, Auto-Attendant 5-15

Auth ID 6-50

AA Dial Plan 5-6, 5-11, 6-18

AA LBR Codec 6-19

Auth Resync-Reboot 6-48

Auto 2-18

AA Preferred Codec 6-19

AA Prompts URL Script 6-20

AA Script 5-6, 5-11, 6-18

AA User Pref Codec Only 6-19

Accept Last Act Code 6-30

Accept Last Deact Code 6-30

accessing IVR 2-8, 2-9

accounts, User and Administrator 2-23

action instruction, Auto-Attendant 5-15

administration privileges 2-24

administration web server

accessing 2-22

Auto-Attendant

action instruction 5-15

administration web server 5-6

architecture 1-12

audio file 5-15

change greetings 2-32

described 1-3

dial plans 5-9

downloading prompts 5-8

forwarding to 4-16

no input instruction 5-15

no match instruction 5-16

parameters 5-11

Auto-Attendant 5-6

cannot access 2-32, 2-33

enabling/disabling 2-11, 2-16

port, checking 2-11

prompts 5-2

scripts 5-10

Setup Wizard 5-4

using 2-22

switching 5-10

Administrator

transferred after one ring 2-32

types 5-1

account name 2-23

Administrator account

using IVR to record prompts 5-2

XML instructions 5-22

for IVR 2-8

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Index

XML scripts 5-13

Call 1 Peer Phone 6-4

Call 1 Remote Hold 6-3

Call 1 Round Trip Delay 6-4

Call 1 State 6-3

automatic provisioning 2-27

AVT Codec Name 6-12

AVT Dynamic Payload 6-11

Call 1 Tone 6-3

Call 1 Type 6-3

Call Back Act Code 6-29

Call Back Active 6-3

B2BUA 4-15

bandwidth budget 2-6

Call Back Busy Act Code 6-29

Call Back Deact Code 6-29

Call Back Delay 6-28

binary format 2-26

Blind Attn-Xfer Enable 6-43

Blind Transfer Code 6-29

Block ANC Act Code 6-31

Block ANC Deact Code 6-31

Block CID Act Code 6-30

Block CID Deact Code 6-31

Block CID Per Call Act Code 6-31

Block CID Per Call Deact Code 6-31

Block Last Act Code 6-30

Block Last Deact Code 6-30

bridge mode 4-33

Call Back Expires 6-28

Call Back Retry Intvl 6-28

Call Capacity 6-50

Callee Conn Polarity 6-46

Callee On Hook Delay 6-28

Caller Conn Polarity 6-46

Caller ID FSK Standard 6-36

Caller ID Method 6-36

call forward bridge mode 4-33

call forwarding 4-17, 4-33

call hunt groups 4-6

Busy Tone 6-23

call management 4-13

Call Park 1-12

call park 4-26

Call 1 Bytes Recv 6-4

Call 1 Bytes Sent 6-4

Call 1 Callback 6-3

Call 1 Decode Latency 6-4

Call 1 Decoder 6-3

Call 1 Duration 6-4

Call 1 Encoder 6-3

Call Park DLG Refresh Intvl 6-15

Call Park MOH Server 6-15

call pickup 4-25

Call Redial Code 6-29

Call Return Code 6-29

Call Routing Rule 4-14, 6-15

call transfer bridge mode 4-33

candidate sequences 4-9

caring for hardware 2-7

Cfwd All Act Code 6-29

Cfwd All Deact Code 6-30

CFWD Bridge Mode 6-56

Cfwd Busy Act Code 6-30

Cfwd Busy Deact Code 6-30

Call 1 FAX 6-3

Call 1 Jitter 6-4

Call 1 Packet Error 6-4

Call 1 Packets Lost 6-4

Call 1 Packets Recv 6-4

Call 1 Packets Sent 6-4

Call 1 Peer Name 6-3

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Index

Cfwd Dial Tone 6-24

Cfwd Last Act Code 6-30

Cfwd Last Deact Code 6-30

Cfwd No Ans Act Code 6-30

Cfwd No Ans Deact Code 6-30

Cfwd No Ans Delay 6-52

change greetings for Auto-Attendant 2-32

check

configuration profile 2-26

Configuration Wizard

Main Menu 2-18

Configure Advanced Features option 2-18

Configure Client Stations option 2-18

Configure SPA400 2-18

Configure SPA9000 option 2-18

Confirm Tone 6-24

administration web server port 2-11

DHCP 2-10

connections

network 2-13

Ethernet port address 2-11

firmware version 2-10

IP address 2-17

physical 2-8

Contact List 4-6, 4-15, 4-16, 5-6, 6-51

CPC Delay 6-29

LAN IP address 2-11

MAC address 2-10

CPC Duration 6-29

Current AA 5-13, 6-4

multicast address 2-11

network mask 2-10

Current Time 6-2

Customization parameter 6-2

CW Act Code 6-30

primary DNS server 2-10

static gateway IP address 2-10

WAN IP address 2-10

CID Act Code 6-31

CWCID Act Code 6-31

CWCID Deact Code 6-31

CW Deact Code 6-30

CWFD Bridge Mode 4-32

CW Per Call Act Code 6-30

CW Per Call Deact Code 6-30

CWT1 Cadence 6-26

CWT2 Cadence 6-26

CWT3 Cadence 6-26

CWT4 Cadence 6-26

CWT5 Cadence 6-26

CWT6 Cadence 6-26

CWT7 Cadence 6-26

CWT8 Cadence 6-26

CWT Frequency 6-27

CID Deact Code 6-31

Client Certificate 6-2

client configuration

local 1-15

manual 2-28

client registration 2-30

client stations

calling between 4-13

configuring 2-18

configuring with Wizard 4-2

firmware 4-3

MAC address 4-3

Conference Act Code 6-32

Conference Bridge Ports 6-43

Conference Bridge URL 6-43

Conference Tone 6-24

configuration

Daylight Saving Time Rule 6-35

Daytime 5-12, 6-18

Daytime AA 5-11, 6-18

saving 2-33

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Daytime AA Script 5-6, 5-11, 6-19

DayTime Answer Delay 5-11, 6-19

debugging 2-33

check 2-10

secondary 2-16

set 2-11

Debug Level 6-6

downloading

Debug Server 6-6

Auto-Attendant prompts 5-8

Configuration Wizard 2-18

Setup Wizard 2-13

Default Group Line 4-14, 6-16

default music file 4-28

Detect ABCD 6-36

DHCP

DTMF digits 1-7, 5-9

DTMFdigits 5-13

check 2-10

DTMF Playback Length 6-36

DTMF Playback Level 6-36

DTMF Process AVT 6-45

DTMF Process INFO 6-45

DTMF Relay MIME Type 6-7

DTMF Tx Method 6-45

enabling/disabling 2-10, 2-15, 2-17

lease renewal 4-34

with Configuration Wizard 2-13

dialogs, Auto-Attendant 5-13

dialplan instruction 5-22

Dial Plan parameter 6-41, 6-53

dial plans 4-9

Auto-Attendant 5-9

entries 4-9

Echo Canc Adapt Enable 6-44

Echo Canc Enable 6-44

Echo Supp Enable 6-44

Elapsed Time 6-2

examples 4-11

functioning 4-14

rules 4-10

Emergency Number 6-43

Enable IP Dialing 6-43

Enable Web Admin Access 6-6

enabling/disabling

statement 5-14

translation 5-10

Dial Tone 6-23

dial tone, in dial plans 4-10

DID

administration server 2-11

administration web server 2-16

DHCP 2-10, 2-15

lines supported by SPA400 2-4

numbers, block of 4-16

Directed Call Pickup 4-25

Direct Inward Dialing

see DID

static IP addressing 2-15

encryption key 2-26

enhancements 4-32

entries

Display Name 6-41, 6-50

Dist Ring Act Code 6-31

Dist Ring Deact Code 6-31

DMTF inputs 5-16

DND Act Code 6-31

DND Deact Code 6-31

DNS server

dial plans 4-9

Escape Display Name 6-8

Ethernet port

check 2-11

SPA400 2-6

examples

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dial plans 4-11

exit 5-23

final forwarding destination 4-19

firewall

extensions

with media proxy 1-7, 2-3

firmware upgrade

hunt groups 4-6

not valid 2-32

remote 2-26

number not assigned 2-31

Phone 1/2 ports 4-4

shared line appearance 4-5, 4-22

external 4-13

firmware version 2-16

check 2-10

client station 4-3

first-time installation 2-3

flash drive 2-6

incoming calls 4-15

external calls

Force G711a Code 6-34

Force G711u Code 6-34

Force G723 Code 6-34

Force G726r16 Code 6-34

Force G726r24 Code 6-34

Force G726r32 Code 6-34

Force G726r40 Code 6-34

Force G729a Code 6-34

Force Media Proxy 1-7, 6-14

formats for profiles 2-26

four-user license 2-13

front panel

problems 2-32

external numbers

calling 4-14

EXT IP 6-14

EXT RTP Port Min 6-14

EXT SIP Port 6-54

EXT SIP Port parameter 6-48

factory defaults

resetting 2-11, 2-16

SPA400 2-6

SPA9000 2-8

SPA9000 2-4

SPA900 series phones 2-16

user 2-11

FXS 1-12

FXS Port Impedance 6-35

FXS Port Input Gain 6-36

FXS Port Output Gain 6-36

FAX CED Detect Enable 6-44

FAX CNG Detect Enable 6-44

FAX Codec Symmetric 6-45

FAX Disable ECAN 6-45

FAX Enable T38 6-45

FAX Line Toggle Code 6-32

fax machine 2-4

G.729 voice codecs 2-6

G711a Codec Name 6-12

G711u Codec Name 6-12

G711u format 4-28

FAX Passthru Codec 6-45

FAX Passthru Method 6-45

FAX Process NSE 6-45

FAX Tone Detect Mode 6-45

Feature Invocation Method 6-36

Feature Invocation Tone 6-24

G723 Codec Name 6-12

G723 Enable 6-44

G726-16 Enable 6-44

G726-24 Enable 6-44

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G726-32 Enable 6-44

G726-40 Enable 6-44

G726r16 Codec Name 6-12

G726r16 Dynamic Payload 6-12

G726r24 Codec Name 6-12

G726r24 Dynamic Payload 6-12

G726r32 Codec Name 6-12

G726r40 6-12

Hook Flash Timer Max 6-28

Hook Flash Timer Min 6-28

Hook Flash Tx Method 6-45

Hook State

6-3

hosted voicemail services 2-3

HTTP 2-26

HTTPS 2-26

G726r40 Codec Name 6-12

G726r40 Dynamic Payload 6-12

G729a Codec Name 6-12

G729a Enable 6-44

Hunt Groups 4-6, 4-14, 6-16

hunt groups 2-18, 4-6, 4-19

G729b Codec Name 6-12

G729b Dynamic Payload 6-12

gateway, SIP-PSTN 1-8

Get ringing calls 4-27

goto 5-23

Idle Polarity 6-46

Immediate reboot 4-27

Immediate restart 4-27

implicit sequences, in dial plans 4-11

imusic 4-28

Graceful reboot 4-27

INFOREQ Dynamic Payload 6-12

initial connection 2-12

Insert VIA received 6-13

Insert VIA rport 6-13

installation

Graceful restart 4-27

greeting 5-2

Group Call Pickup 4-26

Group Page Address 6-14

Group page end 4-27

first time 2-3

Group page start 4-27

group paging 4-27

instructions, XML 5-22

Interactive Voice Response

see IVR

Group x Line 4-15, 6-16

Group x User ID 4-14, 6-15

Interdigit_Long_Timer 4-12

Interdigit_Short_Timeout 4-9

Interdigit_Short_Timer 4-12

Interdigit Long Timer 6-28

Interdigit Short Timer 6-28

interdigit timers 4-11

interfaces

Handle VIA received 6-13

Handle VIA rport 6-13

hardware

care of 2-7

hardware requirements 2-2

Hardware Version 6-2

Holding Tone 6-24

holiday AA 5-10

SPA9000 2-4

internal calls 4-13

Internal Music LBR Codec 6-17

Internal Music Preferred Codec 6-17

Internal Music Refresh Intvl 6-17

Hook Flash MIME Type 6-8

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Internal Music Script 4-28, 6-17

internal music source 4-28

Internal Music URL 4-28, 6-17

Internal Music Use Pref Codec Only 6-18

intersequence tones 4-10

INVITE Expires 6-9

INVITE message 1-7, 4-14

IP address 2-17

License Keys 2-13

Licenses 6-2

licensing 2-13

Line Enable 6-38, 6-47

line interfaces 4-14

Linksys Profile Compiler 2-26

Linksys Voice System (LVS) 1-1

localization 4-7

check 2-17

local timer overrides 4-11

long interdigit timer 4-11

IP multicast 4-27

IP PBX i-xi, 1-1

IVR

accessing 2-9

MAC address

accessing menu 2-8

features depend on configuration 2-9

options 2-10

check 2-10

client stations 4-3

configuring 2-15

password 2-8

MAC Address parameter 6-2

Mailbox Deposit URL 4-18, 6-55

Mailbox ID 6-43

quick reference 1-14, 2-9

time out 2-9

Mailbox Manage URL 6-55

Mailbox Status 6-55

Mailbox Subscribe URL 6-55

Main Menu 2-18

Jitter Buffer Adjustment 6-39

manual client configuration 2-28

manual reboot 2-11

Mapped SIP Port 6-3

match 5-23

key system

SPA400 2-3

SPA 9000 as 2-3

Max Auth 6-7

Max Expires 6-14

Max Forward 6-7

Max Redirection 6-7

Max RTP ICMP Err 6-11

media proxy 1-7, 2-3

menu type dialog 5-13

menu type dialog format 5-14

messages

LAN IP address

check 2-11

Last Called Number 6-3

Last Caller Number 6-3

Last Registration At 6-3

LEDS

group 4-27

SPA9000 2-4

Message Waiting 6-3, 6-41

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Modem Line Toggle Code 6-32

MOH Server 4-28, 6-43

multicast

NSE Dynamic Payload 6-11

number barring 4-10

messages 4-27

Multicast Address 1-13, 4-27, 6-14

multicast address 4-27

check 2-11

obtaining Setup Wizard 2-13

Off Hook Warning Tone 6-23

options

set 2-11

IVR 2-10

music on hold 4-28

music source, internal 4-28

MWI Dial Tone 6-24

Outside Dial Tone 6-23

pagegroup 4-28

paging 4-27

NAT Keep Alive Dest 6-53

NAT Keep Alive Enable 6-53

NAT Keep Alive Intvl 6-14

NAT Keep Alive Msg 6-53

NAT mapping 1-9

parameters

Auto-Attendant 5-11

dial plan 4-9

XML file 2-27

parking lot 4-26

Password 6-50

NAT Mapping Enable 6-53

network address translation 1-9

Network Jitter Level 6-39

network mask

password

IVR 2-8, 2 -12

pause, in dial plans 4-11

Phone 1/2 port

check 2-10

Next Auto User ID 1-15, 2-27, 6-21

Next Registration In 6-3

Nighttime AA 5-10, 5-11, 6-18

Nighttime AA Script 5-11, 6-19

NightTime Answer Delay 5-11, 6-19

node 5-14

assigning extensions 4-4

Phone 1/2 ports 2-4

Phone Config XML 6-22

Phone Dial Plan 4-15, 6-21

Phone Ext Password 6-21

Phone Upgrade Rule 6-21

physical network connections 2-8, 2-13

plain-text file

node type dialog 5-13

noinput 5-22

no input instruction, Auto-Attendant 5-15

nomatch 5-23

profile 2-27

Playback ABCD 6-36

ports

no match instruction, Auto-Attendant 5-16

No Silence Suppression 2-7

NOTIFY messages 4-22

No UDP Checksum 6-11

NSE Codec Name 6-12

SPA9000 2-4

powering on 2-13

Power port

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SPA400 2-6

Prefer G711a Code 6-34

Prefer G711u Code 6-34

Prefer G723 Code 6-34

Prefer G726r16 Code 6-34

Prefer G726r24 Code 6-34

Prefer G726r32 Code 6-34

Prefer G726r40 Code 6-34

Prefer G729a Code 6-34

Preferred Codec 6-44

primary DNS server

check 2-10

Real-Time Protocol 1-6

reboot 2-11

Referee Bye Delay 6-40, 6-49

Referor Bye Delay 6-40, 6 -49

Referral Services Codes 6-32

Refer Target Bye Delay 6-40, 6-49

Refer-To Target Contact 6-41, 6-50

registrar server 1-12

registration 2-13

client 2-30

set 2-11

Reg Max Expires 6-9

privileges, administration 2-24

Product Name 6-2

profile compiler 2-26

profile formats 2-26

Prompt 1 6-4

Reg Min Expires 6-9

Reg Retry Intvl 4-33, 6-9

Reg Retry Intvl Cap 6-10

Reg Retry Long Intvl 4-33, 6-9

Reg Retry Long Random Delay 4-34, 6-9

Reg Retry Random Delay 4-34, 6-9

Reg Retry RSC 4-33

prompts, Auto-Attendant 5-2

Prompt Tone 6-23

provisioning 2-26

ReINVITE Expires 6-9

Release 5.1 enhancements 4-32

Release Unused Codec 6-45

remote provisioning 1-15

remote upgrade 2-26

automatic 2-27

provisioning, remote 1-15

Proxy 6-54

proxy

media 1-7

Remove Last Reg 6-8

SIP 1-6

renew DHCP on SIP request timeout 4-34

Reorder Delay 6-28

Proxy Debug Option 6-15

Proxy Listen Port 6-14

Proxy Network Interface 6-14

PSTN switch 1-8

Reorder Tone 6-23

repetition, in dial plans 4-10

request-URI 4-16

resetting

factory defaults 2-8, 2-11, 2-16

Restricted Access Domains parameter 6-6

Restrict Source IP 6-40, 6-49

resync time 4-8

QoS

switch and router support required 2-2

quick-reference for IVR 1-14, 2-9

Resync URL 2-25

Retry Reg RSC 6-10

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RFC 2543 Call Hold 6-8

Ring1 Cadence 6-25

Ring1 Name 6-26

SAS DLG Refresh Intvl 6-42

SAS Enable 6-42

Ring2 Cadence 6-25

Ring2 Name 6-26

SAS Inbound RTP Sink 6-42

saving configuration 2-33

scenarios

Ring3 Cadence 6-25

Ring3 Name 6-26

call forwarding 4-17

Ring4 Cadence 6-25

Ring4 Name 6-26

script error, XML 5-13

scripts, Auto-Attendant 5-10

scripts, XML 5-13

Ring5 Cadence 6-25

Ring5 Name 6-27

secondary DNS server 2-16

Second Dial Tone 6-23

Secure All Call Act Code 6-31

Secure Call Indication Tone 6-24

Secure No Call Act Code 6-31

Secure One Call Act Code 6-31

Secure One Call Deact Code 6-32

secure provisioning 2-26

Send Resp To Src Port 6-13

sequences, dial plan 4-9

Serial Number 6-2

Ring6 Cadence 6-25

Ring6 Name 6-27

Ring7 Cadence 6-25

Ring7 Name 6-27

Ring8 Cadence 6-25

Ring8 Name 6-27

Ring Back Tone 6-24

Ring Frequency 6-27

ringing once 2-32

ring tone, no 2-32

Ring Voltage 6-27

Ring Waveform 6-27

router, required for connectivity 2-2

routing rule 4-14

Service Annc Base Number 6-33

Service Annc Extension Codes 6-33

Session Initiation Protocol 1-6

set

RTCP Tx Interval 6-11

RTP

multicast address 2-11

network mask 2-10

media proxy 2-3

primary DNS server 2-11

static gateway IP address 2-10

static IP addressing 2-10

Set Local Date (mm/dd) 6-35

Set Local Time (HH/mm) 6-35

setup requirements 2-14

Setup Wizard

RTP CoS Value 6-38, 6-47

RTP Log Intvl 6-40

RTP Log Intvl parameter 6-49

RTP Packet Size 6-10

RTP Port Max 6-10

RTP Port Min 6-10

RTP ToS/DiffServ Value 6-38, 6-47

rules

Auto-Attendant 5-4

configuring features 4-1

obtaining 2-13

dial plans 4-10

use for initial configuration 2-3

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Index

warning 2-14

SIT3 RSC 6-10

Shared Ext 4-22

SIT3 Tone 6-24

shared extensions 2-18

Shared Line 4-22

SIT4 RSC 6-10

SIT4 Tone 6-24

shared line appearance 4-5, 4-22

with key system 2-3

Shared User ID 4-22

short interdigit timer 4-11

Silence Supp Enable 6-44

Silence Threshold 6-44

SIP 100REL Enable 6-48

SIP Accept Language 6-7

SIP CoS Value 6-38, 6-47

SIP Debug Option 6-40, 6-49

SIP DIDN Field 4-16, 6-16

SIP DIDN Param Name 4-16, 6-16

SIP GUID 6-48

software

downloading 2-18

Software Upgrade option 2-18

Software Version 6-2

SPA400

back panel 2-6

front panel 2-6

key system 2-3

ports 2-6

static IP addressing 2-15

SPA400 hardware 2-6

SPA9000

architecture 1-12

localization 4-7

SIP Port 6-39, 6-48

SIP proxy 1-6

SPA900 series phones

DHCP 2-13

SIP Proxy-Require 6-48

SIP-PSTN gateway 1-8

SIP Reg User Agent Name 6-7

SIP Remote-Party-ID 6-39, 6-48

SIP request timeout, DHCP renewal 4-34

SIP Server Name 6-7

SIP T1 6-8

factory defaults 2-16

SPA901 phone

factory defaults 2-16

Space Remaining 6-4

specifies 4-13

Speed Dial Act Code 6-31

static gateway IP address

check 2-10

SIP T2 6-8

SIP T4 6-8

SIP Timer B 6-9

static IP addressing 2-13

check 2-10

SIP Timer D 6-9

SIP Timer F 6-9

enabling/disabling 2-15, 2-17

SPA400 2-15

SIP Timer H 6-9

SIP Timer J 6-9

Stats In BYE 6-11

status, viewing 2-33

Sticky 183 6-41

SIP ToS/DiffServ Value 6-38, 6-47

SIP User Agent Name 6-7

SIT1 RSC 6-10

Streaming Audio Server 2-4

STUN Enable 6-13

STUN Server 6-13

STUN Test Enable 6-13

SIT1 Tone 6-24

SIT2 RSC 6-10

SIT2 Tone 6-24

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Index

sub-sequence in dial plans 4-10

Substitute VIA Addr 6-13

switching between Auto-Attendants 5-10

synchronize time 4-8

syntax

user factory defaults 2-11

User ID 6-41, 6-50

User Password 6-6

dial plans 4-9

variables, in XML files 2-29

viewing status 2-33

VMSP Bridge 6-56

VMWI Refresh Intvl 6-28

voice codecs 2-6

voicemail

plain-text file for profile 2-27

syslog 2-33

Syslog Server parameter 6-6

TFTP 2-26

hosted services 2-3

voice prompts

time out

IVR 2-9

flash drive 2-6

timeout

dial plans 4-9

time synchronization 4-8

Time Zone 6-35

To header 4-16

translation, dial plan 5-10

troubleshooting 2-31

Try Backup RSC 6-10

WAN IP address

check 2-10

web interface URLs 2-24

web server 1-13

Weekend/Holiday AA 5-11, 6-18

Weekend/Holiday AA Script 5-11, 6-19

Weekend/Holiday Answer Delay 6-19

Weekend/Holiday Delay 5-11

weekend AA 5-10

upgrade

remote 2-26

Weekends/Holidays 5-12, 6-19

Upgrade URL 2-25

upgrading software 2-18

URLs

xfer 5-23

administration web server 2-24

USB flash drive 2-6

Use Auth ID 6-50

Use Compact Header 6-8

Use LVS_PROXY 6-22

Use Pref Codec Only 6-44

User account name 2-23

user agent 1-6

XFER Bridge Mode 4-32, 6-5 7

Xfer When Hangup Conf 6-43

XML files

example 2-28

generic.xml 2-29

variables 2-29

XML format 2-26

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Index

XML scripts

Auto-Attendant 5-13

errors 5-13

examples for Auto-Attendant 5-17

instructions for Auto-Attendant 5-22

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