Sipura Technology Sipura SPA-2000 Instruction Manual

© 2003 - 2005 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2)
1
Sipura Technology, Inc.
SPA Administration Guide
July 2005
Version 2.0.10.1

© 2003 - 2005 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2)
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Disclaimer – Please Read:
This document contains implementation examples and techniques using Sipura
Technology, Inc. and, in some instances, other company’s technology and products
and is a recommendation only and does not constitute any legal arrangement
between Sipura Technology, Inc. and the reader, either written or implied. The
conclusions reached and recommendations and statements made are based on
generic network, service and application requirements and should be regarded as a
guide to assist you in forming your own opinions and decision regarding your
particular situation. As well, Sipura Technology reserves the right to change the
features and functionalities for products described in this document at any time.
These changes may involve changes to the described solutions over time.
Use of Proprietary Information and Copyright Notice:
This document contains proprietary information that is to be used only by Sipura
Technology customers. Any unauthorized disclosure, copying, distribution, or use of
this information is prohibited.

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Sipura Technology, Inc.
SPA-2000 Administration Guide
Table of Contents
1. Product Description....................................................................................................................... 7
1.1. Introduction........................................................................................................................... 7
1.2. Large-Scale Deployment of VoIP Endpoints........................................................................ 7
1.2.1. Voice Quality Overview.................................................................................................................. 7
1.3. The Session Initiation Protocol............................................................................................. 9
1.3.1. Why SIP?....................................................................................................................................... 9
1.3.2. Components of a SIP Network......................................................................................................10
1.3.3. Provisioning Overview...................................................................................................................11
1.3.4. Security Overview .........................................................................................................................12
1.3.4.1. Proxy Servers......................................................................................................................13
1.3.5. SIP Services..................................................................................................................................13
1.3.5.1. Basic Services.....................................................................................................................13
1.3.5.2. Enhanced Services..............................................................................................................14
1.3.5.3. PSTN Interworking...............................................................................................................16
1.4. Network Address Translation (NAT) Traversal................................................................... 17
1.4.1. Why NAT?.....................................................................................................................................17
1.4.2. VoIP-NAT Interworking..................................................................................................................17
1.5. SPA Hardware Overview.................................................................................................... 18
2. Installation Overview ................................................................................................................... 20
3. Software Configuration................................................................................................................ 20
3.1. Provisioning ........................................................................................................................ 20
3.1.1. Provisioning Capabilities...............................................................................................................21
3.1.2. Configuration Profile......................................................................................................................21
3.1.3. Provisioning Parameters...............................................................................................................24
3.1.3.1. Firmware Upgrade...............................................................................................................30
3.1.4. Upgrade Parameters.....................................................................................................................31
3.2. Configuration Update.......................................................................................................... 33
3.2.1. Provisioning Server Redundancy..................................................................................................33
3.2.2. SPA Provisioning Flow..................................................................................................................33
3.3. IVR Interface....................................................................................................................... 36
3.4. Web Interface ..................................................................................................................... 40
3.4.1. Web Interface Conventions...........................................................................................................40
3.4.2. Administration Privileges...............................................................................................................40
3.4.3. Basic and Advanced Views...........................................................................................................41
3.4.4. Functional URLs............................................................................................................................41
3.4.4.1. Upgrade URL.......................................................................................................................41
3.4.4.2. Resync URL.........................................................................................................................41
3.4.4.3. Reboot URL.........................................................................................................................42
Through the Reboot URL, you can reboot the SPA................................................................................42
Note: Upon request, the SPA will reboot only when it is idle...................................................................42
3.5. Configuration Parameters................................................................................................... 43
3.5.1. Configuration Profile Compiler ......................................................................................................43
3.5.2. Dial Plan........................................................................................................................................57
3.5.3. System Parameters.......................................................................................................................61
System Configuration ..................................................................................................................................61
Network Configuration.................................................................................................................................61
3.5.4. Provisioning Parameters...............................................................................................................62
3.5.5. Upgrade Parameters.....................................................................................................................63
3.5.6. Protocol Parameters......................................................................................................................63
3.5.6.1. Dynamic Payload Types......................................................................................................66
3.5.6.2. SDP Audio Codec Names....................................................................................................66
3.5.6.3. NAT Support........................................................................................................................67

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3.5.7. Line 1 and Line 2 Parameters .......................................................................................................68
3.5.7.1. User Account Information ....................................................................................................68
3. Restrict Source IP Notes:................................................................................................................. 72
3.5.7.2. Supplementary Services Enable..........................................................................................72
3.5.7.3. Audio Settings......................................................................................................................73
3.5.7.4. Dial Plan ..............................................................................................................................75
3.5.7.5. Polarity Settings...................................................................................................................75
3.5.8. User 1 and User 2 Parameters......................................................................................................75
3.5.8.1. Call Forward And Selective Call Forward/Blocking Settings................................................76
3.5.8.2. Speed Dial Settings .............................................................................................................76
3.5.8.3. Supplementary Service Settings..........................................................................................76
3.5.8.4. Distinctive Ring and Ring Settings.......................................................................................77
3.5.9. Regional Parameters.....................................................................................................................78
3.5.9.1. Call Progress Tones ............................................................................................................78
3.5.9.2. Ring and CWT Cadence......................................................................................................79
3.5.9.3. Control Timer Values (sec)..................................................................................................80
3.5.9.4. Vertical Service Code Assignment.......................................................................................81
3.5.9.5. Outbound Call Codec Selection Codes: ..............................................................................84
3.5.9.6. Secure Call Implementation:................................................................................................85
3.5.9.7. Miscellaneous Parameters...................................................................................................87
3.6. Call Statistics Reporting...................................................................................................... 90
4. SPA-3000 Configuration.............................................................................................................. 92
4.1. Overview............................................................................................................................. 92
4.2. Gateway Call Restriction by Dial Plan................................................................................ 92
4.3. Authentication Methods...................................................................................................... 92
4.4. VoIP-To-PSTN Calls........................................................................................................... 93
4.4.1. One-Stage Dialing.........................................................................................................................93
Range..........................................................................................................................................................93
4.4.2. Two-Stage Dialing.........................................................................................................................94
Range..........................................................................................................................................................94
4.5. PSTN-To-VoIP Calls........................................................................................................... 94
4.5.1. Terminating Gateway Calls ...........................................................................................................95
4.5.2. VoIP Outbound Call Routing .........................................................................................................96
4.6. Failover to PSTN Support................................................................................................... 97
4.7. Line 1 and FXO Sharing One VoIP Account ...................................................................... 97
4.8. PSTN Call to Ring Line 1.................................................................................................... 97
4.9. Symmetric RTP...................................................................................................................98
4.10. Call Progress Tones........................................................................................................... 98
4.10.1. VoIP PIN Tone..........................................................................................................................98
4.10.2. PSTN PIN Tone........................................................................................................................98
4.10.3. Outside Dial Tone.....................................................................................................................98
4.11. Call Scenarios..................................................................................................................... 98
4.11.1. PSTN to VoIP Call w/o Ring-Thru.............................................................................................98
4.11.2. PSTN to VoIP Call w/ Ring-Thru...............................................................................................99
4.11.3. VoIP to PSTN Call by Calling the FXO Number w/ PIN Authentication ....................................99
4.11.4. Line 1 Forward-On-No-Answer to PSTN Gateway.................................................................100
4.11.5. Line 1 Forward-All to PSTN Gateway.....................................................................................100
4.11.6. Line 1 Forward to a Particular PSTN Number ........................................................................100
4.11.7. Line 1 Forward-On-Busy to PSTN Gateway or Number.........................................................100
4.11.8. Line 1 Forward-Selective to PSTN Gateway or Number ........................................................100
4.11.9. Line 1 User Dial 9 to Access PSTN-Gateway for Local Calls.................................................100
4.11.10. Line 1 Uses PSTN-Gateway for 311 and 911 Calls................................................................100
4.11.11. Line 1 Auto-Fallback to PSTN-Gateway.................................................................................101
4.12. PSTN Line Status............................................................................................................. 101
4.13. Summary of SPA-3000 Configuration Parameters........................................................... 103
4.13.1. PSTN Line – Dial Plans..........................................................................................................103
4.13.2. PSTN Line – VoIP-To-PSTN Gateway Setup.........................................................................103
4.13.3. PSTN Line – VoIP Users and Passwords (HTTP Authentication) ..........................................104
4.13.4. PSTN Line – PSTN-To-VoIP Gateway Setup.........................................................................104

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4.13.5. PSTN Line – FXO Timer Values – In seconds .......................................................................106
4.13.6. PSTN Line – PSTN Disconnect Detection..............................................................................106
4.13.7. PSTN Line – International Control..........................................................................................107
4.13.8. Line 1 and PSTN Line – Audio Configuration.........................................................................108
4.13.9. Line 1 – Gateway Accounts....................................................................................................108
4.13.10. Line 1 – VoIP Fallback To PSTN............................................................................................109
4.13.11. Line 1 – Dial Plan ...................................................................................................................109
4.13.12. User1 – Call Forward Settings................................................................................................109
4.13.13. User1 – Selective Call Forward Settings ................................................................................110
4.13.14. Regional – Call Progress Tones.............................................................................................110
4.13.15. Info – FXO Status...................................................................................................................110
4.13.16. PSTN User – PSTN-To-VoIP Selective Call Forward Settings...............................................111
4.13.17. PSTN User – PSTN-To-VoIP Speed Dial Settings.................................................................112
4.13.18. PSTN User – PSTN Ring Thru Line 1 Distinctive Ring Settings.............................................112
4.13.19. PSTN User – PSTN Ring Thru Line 1 Ring Settings..............................................................112
4.13.20. PSTN/VoIP Caller Commands via DTMF...............................................................................112
5. User Guidelines......................................................................................................................... 112
5.1. Basic Services .................................................................................................................. 113
5.1.1. Originating a Phone Call .............................................................................................................113
5.1.2. Receiving a Phone Call...............................................................................................................113
5.2. Enhanced Services........................................................................................................... 113
5.2.1. Caller ID......................................................................................................................................113
5.2.2. Calling Line Identification Presentation (CLIP)............................................................................114
5.2.3. Calling Line Identification Restriction (CLIR) – Caller ID Blocking...............................................114
5.2.4. Call Waiting.................................................................................................................................115
5.2.5. Disable or Cancel Call Waiting....................................................................................................115
5.2.6. Call-Waiting with Caller ID...........................................................................................................116
5.2.7. Voice Mail....................................................................................................................................117
5.2.8. Attendant Call Transfer ...............................................................................................................117
5.2.9. Unattended or “Blind” Call Transfer.............................................................................................118
5.2.10. Call Hold.................................................................................................................................119
5.2.11. Three-Way Calling..................................................................................................................119
5.2.12. Three-Way Ad-Hoc Conference Calling .................................................................................120
5.2.13. Call Return..............................................................................................................................120
5.2.14. Automatic Call Back ...............................................................................................................121
5.2.15. Call FWD – Unconditional ......................................................................................................121
5.2.16. Call FWD – Busy....................................................................................................................122
5.2.17. Call FWD - No Answer ...........................................................................................................123
5.2.18. Anonymous Call Blocking.......................................................................................................124
5.2.19. Distinctive / Priority Ringing and Call Waiting Tone................................................................124
5.2.20. Speed Calling – Up to Eight (8) Numbers or IP Addresses ....................................................125
6. Troubleshooting......................................................................................................................... 125
6.1. Symptoms and Corrections .............................................................................................. 125
6.2. Error and Log Reporting................................................................................................... 125
6.2.1. LED Blink Rate Definitions..........................................................................................................126
7. Feature Descriptions ................................................................................................................. 126
7.1. Data Networking Features................................................................................................ 126
7.1.1. MAC Address (IEEE 802.3).........................................................................................................126
7.1.2. IPv4 – Internet Protocol Version 4 (RFC 791) upgradeable to v6 (RFC 1883)............................126
7.1.3. ARP – Address Resolution Protocol............................................................................................126
7.1.4. DNS – A Record (RFC 1706), SRV Record (RFC 2782).............................................................126
7.1.5. DiffServ (RFC 2475) and ToS – Type of Service (RFC 791/1349)..............................................126
7.1.6. DHCP Client – Dynamic Host Configuration Protocol (RFC 2131)..............................................126
7.1.7. ICMP – Internet Control Message Protocol (RFC792) ................................................................127
7.1.8. TCP – Transmission Control Protocol (RFC793).........................................................................127
7.1.9. UDP – User Datagram Protocol (RFC768)..................................................................................127
7.1.10. RTP – Real Time Protocol (RFC 1889) (RFC 1890)...............................................................127
7.1.11. RTCP – Real Time Control Protocol (RFC 1889)...................................................................127
7.2. Voice Features.................................................................................................................. 127

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7.2.1. SIPv2 – Session Initiation Protocol Version 2 (RFC 3261-3265)................................................127
7.2.1.1. SIP Proxy Redundancy – Static or Dynamic via DNS SRV ...............................................127
7.2.1.2. Re-registration with Primary SIP Proxy Server..................................................................127
7.2.1.3. SIP Support in Network Address Translation Networks – NAT..........................................127
7.2.2. Codec Name Assignment............................................................................................................127
7.2.3. Secure Calls................................................................................................................................127
7.2.4. Voice Algorithms: ........................................................................................................................127
7.2.4.1. G.711 (A-law and mµ-law).................................................................................................128
7.2.4.2. G.726.................................................................................................................................128
7.2.4.3. G.729A ..............................................................................................................................128
7.2.4.4. G.723.1..............................................................................................................................128
7.2.5. Codec Selection..........................................................................................................................128
7.2.6. Dynamic Payload ........................................................................................................................128
7.2.7. Adjustable Audio Frames Per Packet..........................................................................................128
7.2.8. Modem and Fax Pass-Through...................................................................................................128
7.2.9. DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO *)............................................................128
7.2.10. Call Progress Tone Generation..............................................................................................128
7.2.11. Call Progress Tone Pass Through..........................................................................................128
7.2.12. Jitter Buffer – Dynamic (Adaptive)..........................................................................................129
7.2.13. Full Duplex Audio ...................................................................................................................129
7.2.14. Echo Cancellation – Up to 8 ms Echo Tail .............................................................................129
7.2.15. Voice Activity Detection with Silence Suppression & Comfort Noise Generation ...................129
7.2.16. Attenuation / Gain Adjustment................................................................................................129
7.2.17. Signaling Hook Flash Event ...................................................................................................129
7.2.18. Configurable Flash / Switch Hook Timer ................................................................................129
7.2.19. Configurable Dial Plan with Interdigit Timers..........................................................................129
7.2.20. Message Waiting Indicator Tones – MWI...............................................................................130
7.2.21. Polarity Control.......................................................................................................................130
7.2.22. Calling Party Control – CPC...................................................................................................130
7.2.23. International Caller ID Delivery...............................................................................................130
7.2.24. Streaming Audio Server – SAS ..............................................................................................131
7.2.25. Music On Hold – MOH............................................................................................................131
7.3. Security Features.............................................................................................................. 132
7.3.1. Multiple Administration Layers (Levels and Permissions) ...........................................................132
7.3.2. HTTP Digest – Encrypted Authentication via MD5 (RFC 1321) ..................................................132
7.3.3. HTTPS with Client Certificate......................................................................................................132
7.4. Administration and Maintenance Features....................................................................... 132
7.4.1. Web Browser Administration and Configuration via Integral Web Server....................................132
7.4.2. Telephone Key Pad Configuration with Interactive Voice Prompts..............................................132
7.4.3. Automated Provisioning & Upgrade via TFTP, HTTP and HTTPS..............................................132
7.4.4. Periodic Notification of Upgrade Availability via NOTIFY or HTTP..............................................132
7.4.5. Non-Intrusive, In-Service Upgrades ............................................................................................132
7.4.6. Report Generation and Event Logging........................................................................................132
7.4.7. Syslog and Debug Server Records.............................................................................................133
8. Acronyms................................................................................................................................... 133
9. Glossary .................................................................................................................................... 134
10. Index...................................................................................................................................... 136

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1. Product Description
This guide describes basic administration and use of the Sipura Technology SPA-2000 phone
adapter – an intelligent low-density Voice over IP (VoIP) gateway. The SPA-2000 enables carrier
class residential and business IP Telephony services delivered over broadband or high-speed
Internet connections. By intelligent, we mean the SPA-2000 maintains the states of all the calls it
terminates. It is capable of making proper decisions in reaction to user input events (such as on/off
hook or hook flash) with little or no involvement by a ‘middle-man’ server or media gateway controller.
Examples of proper reactions are: playing dial tone, collecting DTMF digits, comparing them against a
dial plan and terminating a call. With intelligent endpoints at the edges of a network, performing the
bulk of the call processing duties, the deployment of a large network with thousands of subscribers
can scale quickly without the introduction of complicated, expensive servers. As described later in
this section, the Session Initiation Protocol (SIP) is a good choice of call signaling protocol for the
implementation of such a device in this type of network.
1.1. Introduction
The phenomenal growth of broadband Internet access (DSL, Cable, FTTH, etc.), has brought the
realization of reliable packet switched IP Telephony Services with circuit switched toll-quality and
subscriber feature transparency with that of the PSTN’s CLASS feature-set. In addition to basic
offerings comparable to traditional PSTN services, many service providers have integrated their IP
Telephony offering with a large number of web-based productivity applications like unified messaging
and call management features such as, remote call forward configuration via the web. Such advances
over traditional phone services, with equal or better voice quality and lower per-minute prices, have
made IP Telephony service a viable business. In fact, IP Telephony service providers in the US and
abroad have seen their subscriber base growing at a rapid pace.
Important!! Please note: The information contained herein is not a warranty from Sipura
Technology, Inc. Customers planning to use the SPA-2000 in a VoIP service deployment are warned
to test all functionality they plan to support in conjunction with the SPA-2000 before putting the SPA-
2000 in service. Some information in Section 1 of this guide is written for educational purposes and
describes functionality not yet implemented in the SPA-2000.
1.2. Large-Scale Deployment of VoIP Endpoints
The technical challenges in deploying and operating a residential IP Telephony service, however, are
not small. One of the main challenges is to make the service transparent to subscribers: The
subscribers shall expect to use their existing phones to make or receive calls in the same way as with
the existing PSTN service. To enable this level of transparency, the IP Telephony solution has to be
tightly integrated. A key element in this end-to-end IP Telephony solution is the provision of an
endpoint device that sits at a subscriber’s premises that serves as an IP Telephony gateway or
telephone adapter. This phone adapter offers one or more standard telephone RJ-11 phone ports –
identical to the phone wall jacks at home – where the subscriber can plug in their existing telephone
equipment to access phone services. The IP Telephony gateway may connect to the IP network, like
the Internet, through an uplink Ethernet connection.
1.2.1. Voice Quality Overview
Voice Quality perceived by the subscribers of the IP Telephony service should be indistinguishable
from that of the PSTN. Voice Quality can be measured with such methods as Perceptual Speech
Quality Measurement (PSQM) (1-5 – lower is better) and Mean Opinion Score (MOS) (1-5 – higher is
better).

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The table below displays speech quality metrics associated with various audio compression
algorithms:
Algorithm Bandwidth Complexity MOS Score
G.711 64 kbps Very Low 4.5
G.726 16, 24, 32, 40 kbps Low 4.1 (32 kbps)
G.729a 8 kbps Low - Medium 4
G.729 8 kbps Medium 4
G.723.1 6.3, 5.3 kbps High 3.8
Please note: The SPA supports all the above voice coding algorithms.
Several factors that contribute to Voice Quality are described below.
Audio compression algorithm – Speech signals are sampled, quantized and compressed before they
are packetized and transmitted to the other end. For IP Telephony, speech signals are usually
sampled at 8000 samples per second with 12-16 bits per sample. The compression algorithm plays a
large role in determining the Voice Quality of the reconstructed speech signal at the other end. The
SPA supports the most popular audio compression algorithms for IP Telephony: G.711 a-law and µ-
law, G.726, G.729a and G.723.1.
The encoder and decoder pair in a compression algorithm is known as a codec. The compression
ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate,
the smaller the bandwidth required to transmit the audio packets. Voice Quality is usually lower with
lower bit rate, however. But Voice Quality is usually higher as the complexity of the codec gets higher
at the same bit rate.
Silence Suppression – The SPA applies silence suppression so that silence packets are not sent to
the other end in order to conserve more transmission bandwidth; instead a noise level measurement
can be sent periodically during silence suppressed intervals so that the other end can generate
artificial comfort noise that mimics the noise at the other end (using a CNG or comfort noise
generator).
Packet Loss – Audio packets are transported by UDP which does not guarantee the delivery of the
packets. Packets may be lost or contain errors which can lead to audio sample drop-outs and
distortions and lowers the perceived Voice Quality. The SPA applies an error concealment algorithm
to alleviate the effect of packet loss.
Network Jitter – The IP network can induce varying delay of the received packets. The RTP receiver
in the SPA keeps a reserve of samples in order to absorb the network jitter, instead of playing out all
the samples as soon as they arrive. This reserve is known as a jitter buffer. The bigger the jitter
buffer, the more jitter it can absorb, but this also introduces bigger delay. Therefore the jitter buffer
size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, then
many late packets may be considered as lost and thus lowers the Voice Quality. The SPA can
dynamically adjust the size of the jitter buffer according to the network conditions that exist during a
call.
Echo – Impedance mismatch between the telephone and the IP Telephony gateway phone port can
lead to near-end echo. The SPA has a near end echo canceller with at least 8 ms tail length to
compensate for impedance match. The SPA also implements an echo suppressor with comfort noise
generator (CNG) so that any residual echo will not be noticeable.

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Hardware Noise – Certain levels of noise can be coupled into the conversational audio signals due to
the hardware design. The source can be ambient noise or 60Hz noise from the power adaptor. The
SPA hardware design minimizes noise coupling.
End-to-End Delay – End-to-end delay does not affect Voice Quality directly but is an important factor
in determining whether subscribers can interact normally in a conversation taking place over an IP
network. Reasonable delay figure should be about 50-100ms. End-to-end delay larger than 300ms is
unacceptable to most callers. The SPA supports end-to-end delays well within acceptable
thresholds.
1.3. The Session Initiation Protocol
1.3.1. Why SIP?
There are many excellent articles and books that discuss the advantages of SIP.1Here are some of
the more popular details:
•SIP message constructs are very similar to those of HTTP which is well-known to be IP
Network (Internet) friendly.
•SIP is transport agnostic – meaning it can be used over TCP/IP or UDP/IP, with or without
security.
•SIP has a better chance of punching through NAT than other control protocols.
•SIP enables the implementation of intelligent endpoints to support scalable advanced
services.
In a nutshell, SIP is a distributed signaling protocol (as opposed to a centralized protocol such as
SS7, MGCP or MEGACO/H.248). With a distributive protocol, the intelligence does not necessarily
reside on a central server, but can be built into the individual endpoints. By moving the intelligence to
reside within the endpoints at the edge of the network, the processing load of the network application
and associated call servers are significantly reduced, thus making the network a very scalable
solution.

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1.3.2. Components of a SIP Network
SIP
Proxy Server
PSTN
Gateway
Router
NAT
Provisioning
Server
PC
PC
Application
Server
ISP
PSTN
Gateway
PSTN
Gateway
Private IP
Network
PSTN
Billing
Server
Subscriber
Database
Subscriber
Domain
Service
Provider
Domain
Application
Server
Broadband
Modem
SP
A
IP
Network
(Internet)
Figure 1 -- Components of a SIP IP Telephony Network
IP Telephony Gateway (SPA): The SPA is a small device that sits at the subscriber’s premises. It
converts between analog telephone signals and IP Telephony signals. It has up to two RJ-11 ports
where standard analog telephones can be directly attached, and an RJ-45 interface for the Ethernet
connection to the home or business LAN. Intelligence can be built into this device to provide a wide
range of features to the subscribers in association with the other elements in the service. The SPA
functions as a SIP User Agent (UA).
Home/SOHO Routers with NAT Functionality: A home/SOHO router is used for routing IP packets
between the subscriber’s private network and the ISP’s public network. If the ISP provides only one
public IP address to the subscriber, the devices attached to the private network will be assigned
private IP addresses and the router will perform network address translation (NAT) on packets sent
from the private network to the public network via the router. Home routers offer the following
features:
•An R-J45 WAN interface for connection to the ISP’s public network and one or more RJ-45
LAN interfaces for connection to the subscriber’s private network. The router directs
packets between the private network and the public network.
•A PPPoE client to connect with the ISP through a DSL modem.
•A DHCP client where the router will obtain an IP address, subnet mask, default router
assignment, etc., for its WAN interface from a DHCP server on the public network.
•A DHCP server for auto-assignment of private IP addresses, subnet mask, and default
router assignment to devices attached to the private network, i.e. computers, IP Telephony

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gateways, etc. The default router in this case is the IP address of the LAN interface of the
router itself.
•Performs NAT on packets sent from the private network to the public network. This is an
important feature such that recipients of the private packets will perceive them as originated
from a public IP address (the router’s WAN interface) and will therefore return messages to
the proper public IP address and port. Different routers may use different rules for
allocating port numbers at the WAN interface to forward packets from a private IP
address/port to a public IP address/port. The allocated port number is also used for routing
packets from external IP addresses to a private address. Most routers will accept a number
of static port mapping rules for forwarding packets received on a specific port at the WAN
interface to a specific IP address/port in the private network.
PSTN - VoIP Gateways: These devices are required if user agents are expected to make calls to or
receive calls from the PSTN. Many gateways may be deployed in order to service a wide area.
Gateways also behave like SIP user agents. The proxy server can be configured with cost-saving
rules based call routing information so that it may decide which gateway to use depending on the
destination and the time of the call. The IP Telephony service provider will assign each subscriber an
E164 telephone number so that it may be reached from the PSTN just like any other telephone.
Billing Servers: Billing servers are used to generate billing data per usage of the IP Telephony
service. Typically, the service provider will charge a flat fee for unlimited calls between IP Telephony
subscribers (on-net-to-on-net calls). Per use or minute chargers will be incurred only when the
subscriber makes calls to PSTN numbers (on-net-to-off-net calls) through one of the PSTN gateways.
CDR (call detail record) data are generated by the PSTN gateway and sent to the billing servers.
Provisioning Servers: Provisioning servers are used to provision the subscriber user agent devices,
e.g. the SPA. When a subscriber signs up for IP Telephony service, he selects an appropriate service
level and enters his personal information including billing information. This information is processed
by the provisioning server and stored into the service provider’s customer database. The provisioning
server generates a device profile based on the subscriber’s choice of options. The device profile,
which is list of configuration parameters, is downloaded into the SPA from the provisioning server.
The SPA can be configured to contact the provisioning server periodically to check for any update of
the device profile, which may include a firmware upgrade or configuration modification to the SPA.
Application Servers: Application servers are used to provide value added services, such as call
forwarding, outgoing or incoming call blocking
Voice Mail Servers: Specialized servers provide voice mail services to the IP Telephony service
subscribers. When the subscriber is busy or the SPA is out of service for maintenance or other
reason, incoming calls to the subscriber may be redirected to the voice mail servers where the caller
can leave a voice mail. The voice mail server will then notify the subscriber’s SPA of the availability of
voice mail(s) in his mailbox. The subscriber can then contact the voice mail server to retrieve his
voice mail(s). The SPA can indicate the message-waiting status to the subscriber through a number
of methods such as stuttered dial tone heard through the telephone every time the subscriber lifts up
the handset until the voice mail is retrieved.
1.3.3. Provisioning Overview
The SPA is configurable in many ways such that it can provide a wide range of customizable services
and operate in many diverse environments with a variety different vendors’ SIP Proxy Servers, VoIP
Gateways, Voice Mail Servers, NAT applications, etc. Provisioning is the process by which the SPA
obtains a set of configuration parameters in order for it to operate in the Service Provider’s network.
The complete set of configuration parameters for an SPA corresponding to an individual subscriber is
referred to as a configuration profile or simply a Profile. The Profile can be encoded as an XML file or
a simple plain text file with a list of tag/value pairs. When the SPA unit is shipped from the factory, it
contains a default common Profile and is considered Unprovisioned. To save costs and expedite

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delivery, however, it is very desirable that an Unprovisioned unit can be shipped directly from the
factory to the subscriber’s location without any preprocessing by the Service Provider.
The SPA contacts the Service Provider’s provisioning server via the IP network or Internet when it is
plugged into the subscriber’s home or business Local Area Network (LAN) – assuming the
provisioning server is reachable from the subscriber’s home network – to pull the designated profile to
be installed in that particular SPA unit. Furthermore, the SPA unit will periodically contact the
provisioning server to download an updated profile. The protocol for downloading the configuration
profile can be “clear text” TFTP or HTTP data or it can be encrypted TFTP, HTTP or HTTPS data if
security is required. Security will be discussed in more details in a later section.
This type of autonomous remote provisioning, where the individual SPA unit pulls the profile from the
provisioning server is very scalable and flexible. Using this provisioning method, a large number of
SPA units can be provisioned simultaneously and updated periodically.
However, some basic information must be provided to the SPA before it can be provisioned in this
fashion: a) the IP address or domain name of the provisioning server to contact, and b) an ID and/or a
password to send to the provisioning server such that it can associate it with a specific subscriber and
obtain the corresponding profile. This information can be sent out-of-band to the subscriber via
secured email or in a letter inside a welcome kit, for example. The subscriber might need to punch in
some numbers using a telephone connected to the SPA in order to enter this information into the unit.
The SPA provides an easy-to-use interface with audio instructions to make this initial configuration
process as painless as possible. An alternative is for the unit to be provisioned with this basic
information by the Service Provider before the unit is shipped to the subscriber.
In addition to the batch mode of remote provisioning, the SPA allows an interactive mode of local
provisioning. One way to offer this feature is through the use of an IVR system (accessed through an
attached telephone set). The user can access a diagnostic or configuration menu to check the status
of the device or to change some of the settings. This method of provisioning may be applied by an
administrator when the device is at the Service Provider’s office, or by the subscriber under the
guidance of trained personnel during over-the-phone troubleshooting.
A third method of entering provisioning information into the SPA is by way of its integral web server
via a browser on a PC. The subscriber has the option to set and adjust configuration parameters via
an easy-to-use, password protected graphical user interface. This method of provisioning might be
preferred by administrators who wish to access the SPA over a secure corporate/institutional LAN or
by the residential subscriber who is a “power user.”
1.3.4. Security Overview
Security may be applied at many levels in the context of the SPA. The following are examples of
information that should be secured:
•The configuration profile pulled from the provisioning server – The downloading of the
profile should be secured since it contains authentication (password/user name ID /
number) information for accessing subscriber telephony services. It may also contain other
passwords and/or encryption keys used for a variety of management and service
operations.
•The administration password to the SPA unit – The unit must disallow access to
administrative functions to unauthorized users. This access can be controlled with an
administrator password. The administrator password can be one of the parameters in the
SPA configuration profile.
•The SIP signaling messages – The SIP messages exchanged between the SIP proxy
server and the SPA should be encrypted with a secret key. This can be achieved, for
instance, by transporting SIP over TLS.

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•RTP packets – The RTP payload exchanged between SIP user agents can be encrypted
with a secret key to protect against eavesdropper. The secret key can be negotiated with
proper SIP signaling messages. Hence the signaling path must be secured also.
1.3.4.1. Proxy Servers
Proxy servers handle two functions:
1. Accept registrations from the SIP user agents,
2. Proxy requests and responses between user agents.
Registration is the process by which a user agent tells the proxy who it is and at what IP address and
port that it can be reached via SIP. Registration usually expires within a finite period (e.g., 60s or
3600s) and the UA shall renew their registration periodically before the last registration expires. When
a user agent initiates a call, it sends a SIP INVITE request to the proxy server and indicates the target
recipient of the call. The proxy server then consults a database to determine where to forward the
request to the destination user agent. The proxy server can request authentication credentials from
the user agent before granting the service. The credentials are computed by the user agent based on
a pre-provisioned password and a challenge “nonce” dynamically generated by the proxy server per
request. This mechanism prevents unauthorized user agents from getting IP Telephony services
through the proxy server. SIP proxy servers are operated by the IP Telephony service provider and
resides at the service provider’s domain. They may be implemented in many different ways. They can
be stateless, stateful, or B2BUA. Stateless proxies do not maintain states of each call; they simply
proxy the requests and responses between the user agents. Hence they are the simplest, most
scalable, but provide the least types of services. Advanced IP Telephony services are possible with
these proxies only with intelligent user agent devices that are capable of delivering these services
without proxy intervention. Stateful proxies maintain the call state of each call and can provide more
intelligent services at the expense of more processing load per call. B2BUA proxies process every
request and response from the user agents and are capable of providing very advance services even
with relatively simple user agent devices. Obviously B2BUA proxies have the highest processing load
per call.
1.3.5. SIP Services
Today’s PSTN offers a large number of enhanced services in addition to basic phone services. Most
of the services offered by the PSTN are accessed by the subscribers through their telephone sets.
The subscribers provide their input by talking into the handset, pressing the keypad, the switch hook
or flash button, while the PSTN presents instructions/information/confirmation to the subscribers
through a variety of audio tones, beeps and/or announcements. The SPA supports a comparable
range of services via a similar user interface in order to make the IP Telephony service transparent to
subscribers.
The SPA is fully programmable and can be custom provisioned to emulate just about any traditional
telephony service available today. This ability to transparently deliver legacy services over an IP
network coupled with the availability of Internet connected devices (PCs. PDA, etc.) and browsers
opens up a new world of potential offerings that a provider can use to differentiate their service and
grow their business.
The following is a list of commonly supported phone services:
1.3.5.1. Basic Services
1.3.5.1.1. Making Calls to PSTN and IP Endpoints
This is the most basic service. When the user picks up the handset, the SPA provides dial tone and is
ready to collect dialing information via DTMF digits from a touch tone telephone. While it is possible to
support overlapped dialing within the context of SIP, the SPA collects a complete phone number and

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sends the full number in a SIP INVITE message to the proxy server for further call processing. In
order to minimize dialing delay, the SPA maintains a dial plan and matches it against the cumulative
number entered by the user. The SPA also detects invalid phone numbers not compatible with the
dial plan and alerts the user via a configurable tone (reorder) or announcement.
1.3.5.1.2. Receiving Calls from PSTN and IP Endpoints
The SPA can receive calls from the PSTN or other IP Telephony subscribers. Each subscriber is
assigned an E.164 phone number so that they may be reached from wired or wireless callers on the
PSTN. The SPA supplies ring voltage to the attached telephone set to alert the user of incoming calls.
1.3.5.2. Enhanced Services
Enhanced Services are provided in addition to Basic calling services and accessed by way of a
touchtone phone through a series of menus. Since the service enabled by the SPA are Internet in
nature, these enhanced services can be made better by offering users a web browser based interface
to control certain aspects of some or all services.
1.3.5.2.1. Caller ID
In between ringing bursts, the SPA can generate a Caller ID signal to the attached phone when the
phone is on-hook.
1.3.5.2.1.1. Calling Line Identification Presentation (CLIP)
Some subscribers will elect to always block their Caller ID information, yet there may be a
circumstance where sending Caller ID information for a particular call is desired, i.e. trying to reach a
party that does not accept Caller ID blocked calls.
The subscriber activates this service to send his Caller ID when making an outgoing call. To activate
the service, the subscriber enters the corresponding * or # code prior to making the call. This service
is in effect only for the duration of the current call.
1.3.5.2.1.2. Calling Line Identification Restriction (CLIR) – Caller ID Blocking
The subscriber activates this service to hide his Caller ID when making an outgoing call. To activate
the service, the subscriber enters the corresponding * or # code prior to making the call. This service
is in effect only for the duration of the current call.
1.3.5.2.2. Call Waiting
The subscriber can accept a call from a 3rd party while engaging in an active call. The SPA shall alert
the subscriber for the 2nd incoming call by playing a call waiting tone.
1.3.5.2.2.1. Disable or Cancel Call Waiting
By setting the corresponding configuration parameter on the SPA, the SPA supports disabling of call
waiting permanently or on a per call basis.
1.3.5.2.2.2. Call-Waiting with Caller ID
In between call waiting tone bursts, the SPA can generate a Caller-ID signal to the attached phone
when it is off hook.
1.3.5.2.3. Voice Mail
1.3.5.2.3.1. Message Waiting Indication
Service Providers may provide voice mail service to their subscribers. When voice mail is available
for a subscriber, a notification message will be sent from the Voice Mail server to the SPA. The SPA
indicates that a message is waiting by, playing stuttered dial tone (or other configurable tone) when
the user picks up the handset.
1.3.5.2.3.2. Checking Voice Mail

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The SPA allows the subscriber to connect to their voice mail box by dialing their personal phone
number.
1.3.5.2.4. Call Transfer
Three parties are involved in Call Transfer: The transferor, transferee, and transfer target. There are 2
flavors of call transfer: Attended Transfer (Transfer with consultation) and Unattended Transfer
(“Blind” Transfer).
1.3.5.2.4.1. Attendant Transfer
The transferor dials the number of the transfer target, then he hangs up (or enters some * or # code)
when the transfer target answers or rings to complete the transfer.
1.3.5.2.4.2. Unattended or “Blind” Transfer
The transferor enters some * or # code and then dials the number of the transfer target to complete
the transfer (without waiting for the target to ring or answer).
1.3.5.2.5. Call Hold
Call Hold lets you put a caller on hold for an unlimited period of time. It is especially useful on phones
without the hold button. Unlike a hold button, this feature provides access to a dial tone while the call
is being held.
1.3.5.2.6. Three-Way Calling
The subscriber can originate a call to a 3rd party while engaging in an active call.
1.3.5.2.7. Three-Way Ad-Hoc Conference Calling
The SPA can host a 3-way conference and perform 3-way audio mixing (without the need of an
external conference bridge device or service).
1.3.5.2.8. Call Return
The SPA supports a service that allows the SPA to automatically dials the last caller’s number.
1.3.5.2.9. Call Return on Busy
If the last called number is busy, the subscriber can order this service to monitor the called party and
to receive a notification from the SPA (such as special phone ring) when that party becomes
available.
1.3.5.2.10. Automatic Call Back
This feature allows the user to place a call to the last number they tried to reach whether the call was
answered, unanswered or busy by dialing an activation code.
1.3.5.2.11. Call Forwarding
These services forward all the incoming calls to a static or dynamically configured destination number
based on three different settings. These services may be offered by the SPA or by the SIP proxy
server. They can be activated by entering certain * or # code, followed by entering a telephone
number to forward calls to. The SPA provides audio instructions to prompt the user for a forwarding
number and confirms that the requested service has been activated.
1.3.5.2.11.1. Call FWD – Unconditional
All calls are immediately forwarded to the designated forwarding number. The SPA will not ring or
provide call waiting when Call FWD – Unconditional is activated.
1.3.5.2.11.2. Call FWD – Busy
Calls are forwarded to the designated forwarding number if the subscriber’s line is busy because of
the following; Primary line already in a call, primary and secondary line in a call or conference.

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1.3.5.2.11.3. Call FWD - No Answer
Calls are forwarded to the designated forwarding number after a configurable time period elapses
while the SPA is ringing and does not answer.
1.3.5.2.12. Anonymous Call Blocking
By setting the corresponding configuration parameter on the SPA, the subscriber has the option to
block incoming calls that do not reveal the caller’s Caller ID.
1.3.5.2.13. Distinctive / Priority Ringing
The SPA supports a number of ringing and call waiting tone patterns to be played when incoming
calls arrive. The choice of alerting pattern to use is carried in the incoming SIP INVITE message
inserted by the SIP Proxy Server (or other intermediate application server in the Service Provider’s
domain).
1.3.5.2.14. Speed Dialing
The SPA supports speed dialing of up to eight (8) phone numbers or IP addresses. To enter a
telephone number speed dial using a touch tone telephone, the user dials a feature code (*74),
followed by a number (2-9), then the destination speed dialed target number. When the user wishes
to speed dial a target number, they press the corresponding speed dial assigned number followed by
the “#” (pound) key.
Users may also enter/review speed dials from User1/User2 web-pages. This interface or similar is
required to enter IP address targets.
1.3.5.3. PSTN Interworking
The SPA is designed to provide a transparent interworking relationship with the PSTN. Service
providers can deploy the SPA in such a way that PSTN endpoints – wired or wireless –
communicating with SPA endpoints do so without modification to their configuration or network
settings.
The service provider may choose to deploy a multi-protocol VoIP network, much the same way the
PSTN supports multiple signaling schemes today. Most telecommunication providers operate
equipment that supports CAS or channel associated signaling, ISDN signaling and SS7 signaling.
When VoIP is introduced or used in the telecommunications landscape, it is likely that the service
provider will implement a signaling gateway that supports multiple IP Telephony protocols along with
legacy PSTN protocols. The signaling gateway is commonly referred to as a Softswitch.
Architecture and functionality can vary greatly amongst the different softswitch vendors. The
protocols used will depend on the types of connections that will be set-up across the service
provider’s network. If the provider is simply providing transport of calls to/from their network to
another provider’s network, but not originating or terminating calls with the endpoints, SIP will likely
be used for softswitch to softswitch communication.
If the service provider is offering origination and/or termination on endpoint equipment then it is very
likely that the softswitch chosen for network operations will support multiple PSTN and VoIP signaling
protocols.
The table below lists the most commonly accepted, de-facto standards used when implementing a
VoIP signaling scheme based on the type of gateway or endpoint equipment being deployed:
VoIP Equipment Type Typical Port Density De-Facto Signaling Standards
Trunking Gateways Greater Than 500 Ports H.248-Megaco / MGCP / IPDC
Access Gateways Between five and 500 Ports SIP / H.323

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PBX/KTS Platforms Between ten and 500 Ports SIP / H.323 / SCCP
PBX/KTS Telephone Sets One Port SIP / MGCP / SCCP
Phone Adapters and IP Centrex
Phones Up to four Ports SIP / MGCP
The SPA supports SIP today. It has the capability to communicate with a variety of endpoints and
signaling entities via SIP messages.
1.4. Network Address Translation (NAT) Traversal
1.4.1. Why NAT?
A NAT allows multiple devices to share the same external IP address to access the resources on the
external network. The NAT device is usually available as one of the functions performed by a router
that routes packets between an external network and an internal (or private) one. A typical application
of a NAT is to allow all the devices in a subscriber’s home network to access the Internet through a
router with a single public IP address assigned by the ISP. The IP header of the packets sent from
the private network to the public network can be substituted by the NAT with the public IP address
and a port selected by the router according to some algorithm. In other words, recipient of the packets
on the public network will perceive the packets as coming from the external address instead of the
private address of the device where the packets are originated.
In most Internet protocols, the source address of a packet is also used by the recipient as the
destination to send back a response. If the source address of the packets sent from the private
network to the public network is not modified by the router, the recipient may not be able to send back
a response to the originator of the message since its private source IP address/port is not usable.
When a packet is sent from a device on the private network to some address on the external network,
the NAT selects a port at the external interface from which to send the packet to the destination
address/port. The private address/port of the device, the external address/port selected by the NAT to
send the packet, and the external destination address/port of the packet form a NAT Mapping.
The mapping is created when the device first sends a packet from the particular source address/port
to the particular destination address/port and is remembered by the NAT for a short period of time.
This period varies widely from vendor to vendor; it could be a few seconds, or a few minutes, or more,
or less. While the mapping is in effect, packets sent from the same private source address/port to the
same public destination address/port is reused by the NAT. The expiration time of a mapping is
extended whenever a packet is sent from the corresponding source to the corresponding destination.
More importantly, packets sent from that public address/port to the external address/port of the NAT
will be routed back to the private address/port of the mapping session that is in effect. Some NAT
devices actually reuse the same mapping for the same private source address/port to any external IP
address/port and/or will route packets sent to its external address/port of a mapping from any external
address/port to the corresponding private source address/port. These characteristics of a NAT can be
exploited by an SPA to let external entities send SIP messages and RTP packets to it when it is
installed on a private network.
1.4.2. VoIP-NAT Interworking
In the case of SIP, the addresses where messages/data should be sent to an SPA are embedded in
the SIP messages sent by the device. If the SPA is sitting behind a NAT, the private IP address
assigned to it is not usable for communications with the SIP entities outside the private network. The
SPA 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 SPA needs to perform the following tasks:

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•Discover the NAT mappings used to communicate with the peer. This could be done with
the help of some external device. For example a server could be deployed on the external
network such that the server will respond to a special NAT-Mapping-Discovery request by
sending back a message to the source IP address/port of the request, where the message
will contain the source IP address/port of the original request. The SPA can send such a
request when it first attempts to communicate with a SIP entity in the public network and
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. Since the mapping is
only alive for short period, the SPA continues to send periodic keep-alive packets through
the mapping to extend its validity as necessary.
1.5. SPA Hardware Overview
The SPA has one of the smallest form factors on the market. It can be installed in minutes as a table-
top or wall mount CPE device. The images below show the SPA-2000. The SPA-1000 and SPA-
3000 are similar to size and shape – the only difference being the color of the adapter.
Figures Figure 2, Figure 3, Figure 4 and Figure 5 show the front, rear, left side and right side of the
SPA-2000, respectively.
Figure 2 – SPA-2000 Front
Figure 3 – SPA-2000 Left Side

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Figure 4 – SPA-2000 Rear
Figure 5 – SPA-2000 Right Side
The SPA has the following interfaces for networking, power and visual status indication:
1. Two (2) RJ-11 Type Analog Telephone Jack Interfaces (Figure 5, above):
These interfaces accept standard RJ-11 telephone connectors. An Analog touchtone telephone or
fax machine may be connected to either interface. If the service supports only one incoming line, the
analog telephone or fax machine should be connected to port one (1) of the SPA. Port one (1) is the
outermost telephone port on the SPA and is labeled “Phone 1.”
The SPA-3000 has an RJ-11 interface labeled “Line” which can be used to connect the adapter to a
PSTN analog telephone circuit.
2. One LED for Unit Status (Figure 5, above):
This LED indicates status via the following behaviors:
ON – LED remains solid on
OFF – LED remains solid off
LONG (Long On) – 3.0s on, 1s off continuously
FAST – 0.1s on, 0.1s off continuously
SLOW – 0.5s on, 0.5s off continuously
VSLO (Very Slow) – 1.0s on, 1.0s off continuously
HB (Heart Beat) – 0.1s on, 0.1s off, 0.1s on, 1s off continuously
HB2 (Heart Beat 2) - 0.1s on, 0.1s off, 0.1s on, 0.1s off, 0.1s on, 1.2s off continuously
ERR0(Error 0) - 0.5s on, 0.3s off, 0.1s on, 0.1s off, 0.1s on, 2s off continuously
ERR1(Error 1) – 0.1s on, 0.1s off, 0.1s on, 0.1s off, 0.5s on, 2s off continuously
ERR2(Error 2) – 0.1s on, 0.1s off, 0.1s on, 0.1s off, 0.5s on, 0.2s off, 0.5s on, 2s off continuously
3. One Ethernet 10baseT RJ-45 Jack Interface (
Figure 3, above):
This interface accepts a standard or crossover Ethernet cable with standard RJ-45 connector. For
optimum performance, Sipura Technology recommends that a Category 5 cable or greater be used in
conjunction with the SPA.

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4. One LED for Data Link and Activity (
Figure 3, above):
This LED indicates status via the following behaviors:
ON – LED remains solid on
OFF – LED remains solid off
FAST – 0.125s on, 0.125s off continuously
SLOW – 0.5s on, 0.5s off continuously
Variable Blink – LED blinks according to packet traffic activity
5. One 5 Volt Power Adapter Interface (
Figure 3, above)
This interface accepts the SPA power adapter that came with the unit. Sipura Technology does not
support the use of any other power adapters other then the power adapter that was shipped with the
SPA unit.
2. Installation Overview
Please check to make sure that you have the following package contents:
1. Sipura Phone Adapter Unit
2. Ethernet Cable
3. RJ-11 Phone Cable (SPA-3000 Only)
4. SPA Quickstart Guide
5. 5 Volt Power Adapter
You will also need:
1. One or Two Analog Touch Tone Telephones (or Fax Machine)
2. Access to an IP Network via an Ethernet Connection
3. Access to a PSTN network connection – SPA-3000 only.
Please observe the following steps to install the SPA.
From the Left Side of the SPA:
1. Insert a standard RJ-45 Ethernet cable (included) into the LAN port.
2. Insert the power adapter cable into the 5V power adapter cable receptacle.
Ensure that the power adapter jack is snugly attached to the SPA.
From the Right Side of the SPA:
1. Insert a standard RJ-11 telephone cable into the Phone 1 port.
2. Connect the other end of the cable to an analog telephone or fax machine.
3. Insert a standard RJ-11 telephone cable into the Phone 2 port (Optional).
4. Connect the other end of the cable to an analog telephone or fax machine.
Note: Do not connect RJ-11 telephone cable from the SPA-1000 or SPA-2000 to the wall jack to
prevent any chance of connection to the circuit switched telco network.
You may now insert the plug end of the power adapter into a live power outlet which will power up the
SPA.