LINKSYS ATA Administrator Guide

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Linksys ATA Administrator Guide

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Linksys ATA Administrator Guide

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Preface

CHAPTER

Document Audience

i-xi

Linksys Analog Telephone Adapters i-xi

How This Document is Organized i-xii

Document Conventions i-xii

Preface

This guide describes administration and use of the Linksys Analog Telephone Adapters (ATAs). It contains the following sections:

• Document Audience, page xi

• Linksys Analog Telephone Adapters, page xi

• How This Document is Organized, page xii

• Document Conventions, page xii

• Related Documentation, page xiii

• Technical Support, page xiii

Document Audience

This document is written for the following audience:

• Service providers offering services using Linksys VoIP products

• VARs and resellers who need configuration information for Linksys VoIP products

• System administrators or anyone who performs Linksys VoIP product installation and

administration

Note This guide does not provide the configuration information required by specific service

providers. Please consult with the service provider for specific service parameters.

Linksys Analog Telephone Adapters

The following summarizes the ports and features provided by the Linksys ATAs described in this document.

• PAP2T—Voice adapter with two FXS ports

• SPA1001—Small VoIP adapter

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• SPA2102—Voice adapter with router

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How This Document is Organized

• SPA3102—Voice adapter with router and PSTN connectivity

• SPA8000—Voice adapter supporting up to eight FXS connections

• AG310—ADSL2+ gateway with VoIP and PSTN connectivity

• RTP300—IP router with two FXS ports

• WRP400—Wireless-G IP router with FXS ports

• WRTP54G—Wireless-G IP router with two FXS ports

• WRT54GP2—Wireless-G IP router with two FXS ports

How This Document is Organized

This document is divided into the following chapters and appendices.

Chapter Contents

Chapter 1, “Introducing Linksys Analog Telephone Adapters”

Chapter 2, “Getting Started” This chapter describes how to use the different administration and

Chapter 3, “Configuring Linksys ATA s ”

Chapter 4, “Configuring the PSTN Gateway (FXO)”

Chapter 5, “Linksys ATA Field Reference”

Appendix A, “Acronyms” This appendix provides the expansion of acronyms used in this

Appendix B, “Glossary” This appendix defines the terms used in this document.

Appendix C, “User Guidelines” This appendix summarizes the operations of ATA user features.

This chapter introduces the Linksys Analog Telephone Adapters (ATAs).

configuration tools provided for managing Linksys ATAs.

This chapter describes how to complete the basic configuration of a Linksys ATA.

This chapter describes how to configure the Linksys SPA3102 and AG310 for providing PSTN connectivity.

This chapter lists the function and usage for each field or parameter on the ATA administration web server pages.

document.

Preface

Document Conventions

The following are the typographic conventions used in this document.

Typographic Element Meaning

Boldface Indicates an option on a menu or a literal value to be entered in a field.

<parameter> Angle brackets (<>) are used to identify parameters that appear on the

Italic Indicates a variable that should be replaced with a literal value.

Monospaced Font Indicates code samples or system output.

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configuration pages of the Linksys ATA administration web server. The index at the end of this document contains an alphabetical listing of each parameter, hyperlinked to the appropriate table in

Field Reference”

Chapter 5, “Linksys ATA

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Preface

Technical Support

Introducing Linksys Analog Telephone Adapters

This guide describes the administration and use of Linksys analog telephone adapters (ATAs). This chapter introduces the functionality of the Linksys ATAs and includes the following sections:

• Overview, page 1-1

• Ensuring Voice Quality, page 1-3

• Feature Descriptions, page 1-5

• Technology Background, page 1-9

• Where to Go From Here, page 1-13

Overview

Table 1-1 summarizes the ports and features provided by the Linksys ATAs described in this document.

Table 1-1 Linksys ATAs

CHA PTER

FXS Product Name

PAP2T Two (2) — One (1) — Two (2) Voice adapter with two FXS ports

SPA1001 One (1) — One (1) — One (1) Small VoIP adapter

SPA2102 Two (2) — One (1) — One (1) Voice adapter with router

SPA3102 One (1) One (1) One (1) One (1) Two (2) Voice adapter with router and PSTN

SPA8000 Eight (8) — One (1) Maintenance

RTP300 Two (2) — One (1) Four (4) Two (2) IP router with two FXS ports

WRP400 WRTP54G WRT54GP2

AG310 One (1) One (1) One (1) One (1) One (1) ADSL2+ gateway with VoIP and

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(Analog

Phone)

Two (2) — One (1) Four (4) Two (2) Wireless-G IP router with two FXS

FXO PSTN Connection

RJ-45 Internet (WAN)

RJ-45 Ethernet (LAN)

only

Configurable Voice Lines

Four (4) Voice adapter with support for up to

Description

connectivity

eight FXS devices

ports

PSTN connectivity

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Overview

Chapter 1 Introducing Linksys Analog Telephone Adapters

Figure 1-1 illustrates how the different ATAs provide voice connectivity in a VoIP network, including

the SPA3102, which acts as a SIP-PSTN gateway. As shown, the following devices also provide QoS-enabled IP routers in addition to ports for connecting analog telephone devices:

• WRP400

• RTP300

• WRTP54G

• WRTP54GP2

Figure 1-1 Linksys ATAs in the VoIP Network and PSTN

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The AG310 and SPA3102 provide full PSTN connectivity in addition to a single FXS port. In addition, the AG310 provides an ADSL2+ gateway.

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Chapter 1 Introducing Linksys Analog Telephone Adapters

Each Linksys ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP Telephony services delivered over broadband or high-speed Internet connections. Linksys ATAs maintain the states of all the calls it terminates and makes the proper reaction to user input events (such as on/off hook or hook flash). Because the ATAs use the SIP standard, there is little or no involvement by a “middle-man” server or media gateway controller.

The response of a Linksys ATA includes playing a dial tone, collecting DTMF digits and comparing them against a dial plan, or terminating the call.

Note The information contained in this guide is not a warranty from Linksys, a division of Cisco Systems, Inc.

Customers planning to use Linksys ATAs in a VoIP service deployment are advised to test all functionality they plan to support before putting the ATA in service.

By implementing Linksys ATAs with the SIP protocol, intelligent endpoints at the edges of a network perform the bulk of the call processing. This allows the deployment of a large network with thousands of subscribers without complicated, expensive servers.

The ATA is a key element in the end-to-end IP Telephony solution. It provides one or more standard telephone RJ-11 phone ports (identical to the telephone phone wall jacks) to which the subscriber connects standard analog telephone equipment to access phone services. The ATA connects to a wide area IP network, such as the Internet, through a broadband (DSL or cable) modem or router.

Ensuring Voice Quality

Voice quality, as perceived by the subscribers of the IP Telephony service, should be equivalent (or better) compared to the PSTN. Voice quality can be measured with such methods as Perceptual Speech Quality Measurement (PSQM), with a scale of 1–5, in which lower is better; and Mean Opinion Score (MOS), with a scale of 1–5, in which higher is better.

Table 1-2 displays speech quality metrics associated with various audio compression algorithms.

Table 1-2 Speech Quality Metrics

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

Note The Linksys ATA supports all the above voice coding algorithms.

The following sections describe the factors that contribute to voice quality.

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Ensuring Voice Quality

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 Linksys ATA 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. Although voice quality is usually lower with a lower bit rate, it is usually higher as the complexity of the codec gets higher at the same bit rate.

Silence Suppression

The Linksys ATA applies silence suppression so that silence packets are not sent to the other end 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).

Chapter 1 Introducing Linksys Analog Telephone Adapters

Packet Loss

Network Jitter

Echo

Audio packets are transported by UDP, which does not guarantee the delivery of the packets. Packets may be lost or contain errors that can lead to audio sample drop-outs and distortions and lowers the perceived voice quality. The Linksys ATA applies an error concealment algorithm to alleviate the effect of packet loss.

The IP network can induce varying delay of the received packets. The RTP receiver in the Linksys ATA keeps a reserve of samples 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 the jitter buffer size is too small, late packets may be considered lost and this lowers the voice quality. The Linksys ATA can dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call.

Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. The Linksys ATA has a near-end echo canceller with at least 8 ms tail length to compensate for impedance match. The Linksys ATA also implements an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable.

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Chapter 1 Introducing Linksys Analog Telephone Adapters

Hardware Noise

Certain levels of noise can be coupled into the conversational audio signals because of the hardware design. The source can be ambient noise or 60 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 50–100 ms. End-to-end delay larger than 300 The Linksys ATA supports end-to-end delays well within acceptable thresholds.

Feature Descriptions

The Linksys ATA is a full featured, fully programmable phone adapter that can be custom provisioned within a wide range of configuration parameters. This chapter contains a high-level overview of features to provide a basic understanding of the feature breadth and capabilities of the Linksys ATA.

• SIP Proxy Redundancy, page 1-5

• Supported Codecs, page 1-6

Feature Descriptions

Hz noise from the power adaptor. The Linksys ATA

ms is unacceptable to most callers.

• Streaming Audio Server and Music on Hold, page 1-6

• Silence Suppression and Comfort Noise Generation, page 1-7

• Modem and Fax Pass-Through, page 1-7

• Adaptive Jitter Buffer, page 1-7

• Other Features, page 1-8

SIP Proxy Redundancy

In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. An average SIP proxy server may handle tens of thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. The Linksys ATA supports the use of backup SIP proxy servers so that service disruption should be nearly eliminated.

A relatively simple way to support proxy redundancy is to configure a static list of SIP proxy servers to the Linksys ATA in its configuration profile where the list is arranged in some order of priority. The Linksys ATA attempts to contact the highest priority proxy server whenever possible. When the currently selected proxy server is not responding, the Linksys ATA automatically retries the next proxy server in the list.

The dynamic nature of SIP message routing makes the use of a static list of proxy servers inadequate in some scenarios. In deployments where user agents are served by different domains it is not feasible to configure a static list of proxy servers for each domain.

One solution in this situation is through the use of DNS SRV records. The Linksys ATA can be instructed to contact a SIP proxy server in a domain named in the SIP message. The Linksys ATA consults the DNS server to get a list of hosts in the given domain that provides SIP services. If an entry

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Feature Descriptions

exists, the DNS server returns an SRV record that contains a list of SIP proxy servers for the domain, with their host names, priority, listening ports, and so on. The Linksys ATA tries to contact the list of hosts in the order of their stated priority.

If the Linksys ATA is currently using a lower priority proxy server, it periodically probes the higher priority proxy to see whether it is back on line, and switches back to the higher priority proxy when possible.

Supported Codecs

Negotiation of the optimal voice codec sometimes depends on the ability of the Linksys ATA device to match a codec name with the codec used by the far-end device. The Linksys ATA allows the network administrator to individually name the various codecs that are supported so that the Linksys ATA can successfully negotiate the codec with the far-end equipment. The administrator can select which low-bit-rate codec is to be used for each line. G.711a and G.711u are always enabled.

Table 1-3 Codecs Supported by Linksys ATAs

Codec (Voice Compression Algorithm)

G.711 (A-law and mµ-law) This very low complexity codec supports uncompressed 64 kbps

G.726 This low complexity codec supports compressed 16, 24, 32, and

G.729A The ITU G.729 voice coding algorithm is used to compress

G.723.1 The Linksys ATA supports the use of ITU G.723.1 audio codec at

Chapter 1 Introducing Linksys Analog Telephone Adapters

Description

digitized voice transmission at one through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs.

40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.

digitized speech. Linksys supports G.729. G.729A is a reduced complexity version of G.729. It requires about half the processing power to code G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not identical.

6.4 kbps. Up to two channels of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1 simultaneously, or Line 1 or Line 2 can initiate a three-way conference with both call legs using G.723.1.

Note When no static payload value is assigned per RFC 1890, the Linksys ATA can support dynamic payloads

for G.726.

Streaming Audio Server and Music on Hold

This feature allows you to attach an audio source to one of the Linksys ATA FXS ports and use it as a streaming audio source device. The corresponding Line (1 or 2) can be configured as a streaming audio server (SAS) such that when the Line is called, the Linksys ATA answers the call automatically and

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starts streaming audio to the calling party provided the FXS port is off-hook. If the FXS port is on-hook when the incoming call arrives, the Linksys ATA replies with a SIP 503 response code to indicate “Service Not Available.”

If an incoming call is auto-answered, but later the FXS port becomes on-hook, the Linksys ATA 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 Linksys ATA replies with a SIP 486 response code to indicate “Busy Here”. The SAS line can be set up to refresh each streaming audio session periodically (via SIP re-INVITE) to detect whether 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.

On a connected call, the Linksys ATA may place the remote party on call. The only way to do this on the SPA2102 is to perform a hook-flash to initiate a three-way call, or to swap two calls during call-waiting. If the remote party indicates that they can still receive audio while the call is holding, the SPA2102 can be set up to contact an auto-answering SAS as described in and have it stream audio to the holding party. When used this way, the SAS is referred to as a MOH Server.

Silence Suppression and Comfort Noise Generation

Feature Descriptions

Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls supported by the network by reducing the required bidirectional bandwidth for a single call. VAD uses a very sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present. If the VAD algorithm decides speech is not present, the silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network.

Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to reassure callers that their calls are still connected during silent periods. If Comfort Noise Generation is not used, the caller may think the call has been disconnected because of the “dead silence” periods created by the VAD and Silence Suppression feature.

Modem and Fax Pass-Through

Modem Pass-through Mode can be triggered only by predialing the <Modem Line Toggle Code>. FAX Pass-through Mode is triggered by CED/CNG tone or NSE events. Echo canceller is automatically disabled for Modem Pass-through Mode only. Echo canceller is automatically disabled only if <FAX Disable ECAN> (Line 1/2) is set to “yes” for that line (in that case FAX pass-through is the same as Modem pass-through). Call waiting and silence suppression is automatically disabled for both FAX and Modem pass-through as before. In addition, out-of-band DTMF Tx is disabled during modem or fax pass-through.

Adaptive Jitter Buffer

The Linksys ATA can buffer incoming voice packets to minimize out-of-order packet arrival. This process is known as jitter buffering. The jitter buffer size proactively adjusts or adapts in size, depending on changing network conditions.

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Feature Descriptions

Other Features

Chapter 1 Introducing Linksys Analog Telephone Adapters

The Linksys ATA has a Network Jitter Level control setting for each line of service. The jitter level decides how aggressively the Linksys ATA tries to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly.

The following table summarizes other features provided by Linksys ATAs.

Table 1-4 Linksys ATA Features

Feature Description

International Caller ID Delivery

Secure Calls A user (if enabled by service provider or administrator) has the option to

Adjustable Audio Frames Per Packet

DTMF In-Band and Out-of-Band (RFC 2833) (SIP INFO *) The Linksys ATA

Call Progress Tone Generation

Call Progress Tone Pass Through

Full Duplex Audio Full-duplex is the ability to communicate in two directions simultaneously

Echo Cancellation—Up to 8 ms Echo Tail

In addition to support of the Bellcore (FSK) and Swedish/Danish (DTMF) methods of Caller ID (CID) delivery, release 2.0 adds a large subset of ETSI-compliant methods to support international CID equipment. Different types of CID delivery method can be obtained by mixing-and-matching some of the steps as shown.

make an outbound call secure in the sense that the audio packets in both directions are encrypted.

This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality.

may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information.

The Linksys ATA has configurable call progress tones. Parameters for each type of tone may include number of frequency components, frequency and amplitude of each component, and cadence information.

This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network.

so that more than one person can speak at a time. Half-duplex means that only one person can talk at a time, like a CB radio or walkie-talkie, which is unnatural in normal free-flowing two-way communications. The Linksys ATA supports full-duplex audio.

The SPA3102 supports hybrid line echo cancellation. This feature uses the G.165 echo canceller to eliminate up to 8 ms of line echo. This feature does not provide acoustic echo cancellation on endpoint devices; that is, an end user speakerphone.

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Chapter 1 Introducing Linksys Analog Telephone Adapters

Table 1-4 Linksys ATA Features

Feature Description

Signaling Hook Flash Event

Configurable Dial Plan with Interdigit Timers

Technology Background

The Linksys ATA can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-call-control. Depending on the features that the service provider offers using third-party-call-control, the following three Linksys ATA features may be disabled to correctly signal a hook-flash event to the softswitch:

• Call Waiting Serv

• Three Way Call Serv

• Three Way Conf Serv

You can configure the length of time allowed for detection of a hook flash using the Hook Flash Timer parameter on the Regional tab of the administration web server.

The Linksys ATA has three configurable interdigit timers:

• Initial timeout (T)—Handset off hook, no digit pressed yet.

• Long timeout (L)—One or more digits pressed, more digits needed to

reach a valid number (as per the dial plan).

Polarity Control The Linksys ATA allows the polarity to be set when a call is connected and

Calling Party Control Calling Party Control (CPC) signals to the called party equipment that the

Report Generation and Event Logging

Syslog and Debug Server Records

Technology Background

• Short timeout (S)—Current dialed number is valid, but more digits

would also lead to a valid number.

when a call is disconnected. This feature is required to support some pay phone system and answering machines.

calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily. This feature is useful for auto-answer equipment, which then knows when to disengage.

The Linksys ATA reports a variety of status and error reports to assist service providers to diagnose problems and evaluate the performance of their services. The information can be queried by an authorized agent, using HTTP with digested authentication, for instance. The information may be organized as an XML page or HTML page.

The Linksys ATA supports detailed logging of all activities for further debugging. The debug information may be sent to a configured Syslog server. Via the configuration parameters, the Linksys ATA allows some settings to select which type of activity/events should be logged, as for instance, a debug level setting.

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This section provides background information about the technology and protocols used by the ATA. It includes the following topics:

• Session Initiation Protocol, page 1-10

• Network Address Translation, page 1-10

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Technology Background

Session Initiation Protocol

Linksys ATAs are implemented using open standards, such as Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. to another subscriber in the network. The requestor is called the user agent server (UAS), while the recipient is called the user agent client (UAC).

Figure 1-2 SIP Requests and Responses

SIP UA

RTP

Chapter 1 Introducing Linksys Analog Telephone Adapters

Figure 1-2 illustrates a SIP request for connection

SIP Proxy

SIP UA

In a SIP VoIP network, when the SIP proxy receives a request from a 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).

Network Address Translation

This section describes issues that arise when using a Linksys ATA on a network behind a network address translation (NAT) device. It includes the following topics:

• NAT Overview, page 1-10

• NAT Types, page 1-11

• Simple Traversal of UDP Through NAT, page 1-12

• SIP-NAT Interoperation, page 1-12

SIP Proxy

NAT Overview

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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.

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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 1-3 NAT Support with Session Border Controller Provided by ITSP

Technology Background

NAT Types

Private IP address

192.168.1.1

Linksys ATA

192.168.1.100

NAT Device

DHCP server

External IP address assigned by ISP

ISP

Internet

ITSP

Session Border

Controller

The ITSP may support NAT mapping using a Session Border Controller (see Figure 1-3). This is the preferred option because it eliminates the need for managing NAT on the Linksys ATA. If this is not available, you will need to discuss with the ITSP how to use the NAT Support Parameters provided by the Linksys ATA, such as <Outbound Proxy> and <STUN Server Enable>.

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The different ways that NAT is implemented is 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.

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Technology Background

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, 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

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 Linksys ATA:

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.

Chapter 1 Introducing Linksys Analog Telephone Adapters

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

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

Step 3 View the syslog messages to determine if your network uses symmetric NAT or not.

SIP-NAT Interoperation

In the case of SIP, the addresses where messages/data should be sent to a Linksys ATA system are embedded in the SIP messages sent by the device. If the Linksys ATA 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 Linksys ATA 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 Linksys ATA system needs to perform the following tasks:

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• 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

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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 Linksys ATA 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 Linksys ATA system continues to send periodic keep-alive packets through the mapping to extend its validity as necessary.

Where to Go From Here

To Refer to

Use the different administration and

Chapter 2, “Getting Started”

configuration tools provided for managing Linksys ATAs.

Complete the basic configuration of a Linksys

Chapter 3, “Configuring Linksys ATAs”

ATA.

Configure the Linksys SPA3102 or AG310 for providing PSTN connectivity.

Look up the function and usage for each field or

Chapter 4, “Configuring the PSTN Gateway (FXO)”

Chapter 5, “Linksys ATA Field Reference”

parameter on the ATA administration web server pages.

Look up the expansion of acronyms used in this

Appendix A, “Acronyms”

document.

Define the terms used in this document. Appendix B, “Glossary”

Understand the operations of ATA user features. Appendix C, “User Guidelines”

The following documentation provides additional information about features and functionality of Linksys ATAs:

• AA Quick Guide

• IVR Quick Guide

• SPA Provisioning Guide

The following documentation describes how to use and configure other Linksys VoIP products:

Document Version 3.1

• SPA9000 Administrator Guide

• LVS CTI Integration Guide

• LVS Integration with ITSP Hosted Voicemail Guide

Linksys ATA Administrator Guide

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Where to Go From Here

Chapter 1 Introducing Linksys Analog Telephone Adapters

• SPA900 Series IP Phones Administrator Guide

• Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment

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CHA PTER

Getting Started

This chapter provides a brief description of each Linksys ATA and describes the tools and utilities available for administration. It includes the following sections:

• Linksys Analog Telephone Adapters (ATAs), page 2-1

• Establishing Connectivity, page 2-14

• Connecting the SPA8000, page 2-16

• Using the Interactive Voice Response Interface, page 2-18

• Using the Administration Web Server, page 2-22

The following ATAs provide Layer 2 (data link) and Layer 3 (IP) configuration options:

• AG310

• WRP400

• WRTP54/54G

• RTP300

For information about configuring these options, including private virtual circuits (PVCs), PPPoE, and IP routing, refer to the user guide for the specific device.

For information about configuring the Public Switch Telephone Network (PSTN) connectivity options for the SPA3102 and AG310 refer to

Chapter 3, “Configuring Linksys ATAs.”

Linksys Analog Telephone Adapters (ATAs)

Linksys ATAs convert voice traffic into data packets for transmission over an IP network. This section illustrates and summarizes the ports and LEDs provided by each model. It includes the following topics:

• Caring for Your Hardware, page 2-2

• PAP2T, page 2-3

• SPA2102, page 2-6

• SPA3102, page 2-7

• SPA8000, page 2-9

• AG310, page 2-2

• WRP400, page 2-10

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Linksys Analog Telephone Adapters (ATAs)

• WRTP54G, page 2-11

• WRT54GP2, page 2-13

• RTP300, page 2-4

Caring for Your Hardware

The Linksys ATA is an 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.

AG310

The AG310 provides a high-speed ADSL2/+ modem along with a four-port Ethernet switch and a PSTN gateway (see

Note Througout this document, when references are made to the software configuration of the SPA3102, the

information also applies to the AG310, which provides the same functionality as the SPA3102, as well as an ADSL modem and a four-port Ethernet switch.

Figure 1-4).

Chapter 2 Getting Started

Figure 1-4 AG310

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

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Table 1-5 AG310 Front Panel

LED Function

Power Steady green indicates power on and Internet connection. Flashing indicates

not connected to the Internet, booting or firmware upgrade.

Ethernet 1-4 Steady green indicates an active connection to the network. Flashing

indicates traffic.

Voice Status Steady green indicates that a voice call is currently in progress.

Phone Steady green indicates active or registered connection to the ITSP through

the Phone port. Flashing indicates device is in use or off hook.

Line Steady green when a telephone or fax machine has an active connection to

traditional phone service through the Line port.

DSL Steady green indicates that the DSL connection is active.

Internet Steady green indicates active connection. Flashing indicates traffic.

Table 1-6 AG310 Back Panel

PAP2T

Port Function

DSL Connect to the telephone wall jack providing DSL service using a RJ-11

cable.

Line Connect to a standard telephone wall jack with an RJ-11cable.

Phone Connect to an analog telephone or fax machine with an RJ-11 cable.

Ethernet 1-4 Connect to network devices, such as a PC or a switch, using an Ethernet

cable.

Power Connect to the power supply.

The PAP2T provides two FXS ports (see Figure 1-5).

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Figure 1-5 PAP2T

Chapter 2 Getting Started

RTP300

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-7 PAP2T Front Panel

LED Function

Phone 1/2 Steady green indicates active or registered connection to the ITSP through

the Phone port. Flashing indicates device is in use or off hook.

Ethernet Steady green indicates active connection. Flashing indicates traffic.

Power Steady green indicates power on and Internet connection. Flashing indicates

not connected to the Internet, booting or firmware upgrade.

Table 1-8 PAP2T Back Panel

Port Function

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

Ethernet Connect to a router or broadband (cable/DSL) modem.

Power Connect to the 5-volt DC power adapter.

The RTP300 provides an ATA with two FXS ports along with an IP router (see Figure 1-6). The RTP300 provides connectivity to an analog telephone as well as Internet connectivity to a LAN with a built-in four-port switch. The IP router provides QoS support, an integrated firewall and VPN pass-through, and full routing with DHCP and NAT.

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Linksys Analog Telephone Adapters (ATAs)

Figure 1-6 RTP300

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-9 RTP300 Front Panel

LED Function

Ethernet 1-4 Steady green indicates an active connection to the network. Flashing

indicates traffic.

Phone 1/2 Steady green when telephone or fax machine has an active or registered

connection to the ITSP through the Phone port. Flashing indicates phone is in use or off hook.

Internet Steady green indicates active connection. Flashing indicates traffic.

Table 1-10 RTP300 Back Panel

Port Function

Internet Connect to the broadband (cable/DSL) modem using an Ethernet cable.

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

Ethernet 1-4 Connect to local IP devices, such as PCs, using an Ethernet cable.

Power Connect to a 12v DC power supply.

SPA1001

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The SPA1001 provides one FXS port (see Figure 1-7).

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Linksys Analog Telephone Adapters (ATAs)

Figure 1-7 SPA1001

The following table describes the LEDS and ports on the back panel of the device.

Table 1-11 SPA1001 Back Panel

LED/Port Function

Phone Connect to an analog telephone or fax machine with an RJ-11 cable.

Power Connect to the 5-volt DC power adapter.

Act Steady green indicates active or registered connection to the ITSP through

Ethernet Connect to a router or broadband (cable/DSL) modem.

Chapter 2 Getting Started

the Phone port. Flashing indicates device is in use or off hook.

Steady yellow indicates active Ethernet connection. Flashing indicates traffic.

SPA2102

The SPA2102 provides two FXS ports and an Ethernet port for connecting to a local PC or other Ethernet device.

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Figure 1-8 SPA2102

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-12 SPA2102 Front Panel

LED Function

Power Steady green indicates power on and Internet connection. Flashing indicates

not connected to the Internet, booting or firmware upgrade.

Internet Steady green indicates active connection. Flashing indicates traffic.

Phone 1/2 Steady green indicates active or registered connection to the ITSP through

the Phone port. Flashing indicates device is in use or off hook.

Table 1-13 SPA2102 Back Panel

Port Function

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

Ethernet Connect to a network device, such as a PC or a switch with an Ethernet

cable.

Internet Connect to a router or broadband (cable/DSL) modem.

Power Connect to the 5 volt DC power adapter.

SPA3102

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The SPA3102 acts as a PSTN gateway providing connectivity to the PSTN.

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Figure 1-9 SPA3102

Chapter 2 Getting Started

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-14 SPA3102 Front Panel

LED Function

Power Steady green indicates power on and Internet connection. Flashing indicates

not connected to the Internet, booting or firmware upgrade.

Internet Steady green indicates active connection. Flashing indicates traffic.

Phone Steady green indicates active or registered connection to the ITSP through

the Phone port. Flashing indicates device is in use or off hook.

Line Steady green when a telephone or fax machine has an active connection to

traditional phone service through the Line port.

Table 1-15 SPA3102 Back Panel

Port Function

Line Connect to a standard telephone wall jack with an RJ-11cable.

Phone Connect to an analog telephone or fax machine with an RJ-11 cable.

Ethernet Connect to a network device, such as a PC or a switch with an Ethernet

cable.

Internet Connect to a router or broadband (cable/DSL) modem.

Power Connect to the power supply.

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SPA8000

Linksys Analog Telephone Adapters (ATAs)

The SPA8000 is an analog telephone adapter that supports connecting up to eight analog telephone devices, including up to four fax devices. The SPA8000 consists of four hardware modules, each of which is similar in functionality to the SPA2102. However, IP routing functionality is not provided by the SPA8000.

Figure 1-10 SPA8000

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-16 SPA8000 Front Panel

LED Function

Power Steady green indicates power on and Internet connection. Flashing indicates

not connected to the Internet, booting or firmware upgrade.

Ethernet Link activity status. Steady green indicates active connection. Flashing

indicates traffic.

Status 1/2/3/4 Indicates the status of each of the four hardware modules in the SPA8000.

Steady green indicates active connection. Flashing indicates tht the associated module is restarting.

Line 1/2/3/4/5/6/7/8 Steady green indicates active or registered connection to the ITSP. Flashing

indicates device is in use or off hook.

Table 1-17 SPA8000 Back Panel

Port Function

RJ21 port Provides a single connector that supports up to eight FXS connections.

RJ11 ports: Line 1/2 Line 3/4 Line 5/6

Connect each of the eight FXS ports to an analog telephone or fax machine using an RJ-11 cable. These ports are distributed among four hardware modules of two ports each. You can connect a single fax machine to each pair of ports (for a maximum of four).

Line 7/8

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Table 1-17 SPA8000 Back Panel

Port Function

AUX Ethernet maintenance port. Connect to a network device, such as a PC or a

Ethernet Connect to a switch, router or broadband (cable/DSL) modem for access to

Power Connect to the 12-volt DC power adapter.

Chapter 2 Getting Started

switch with an Ethernet cable for accessing the administration web server on the SPA8000. The attached device is assigned an IP address dynamically by the internal DHCP server provided by the SPA8000.

the Internet.

Note When connected directly to the Internet, connectivity between FXS

devices on the SPA8000 depends on the Outbound Proxy service provided by the ITSP.

Caution This 12-volt power adapter is to be used only with the SPA8000.

Using it with any other Linksys product may damage the unit.

WRP400

The WRP400 provides an ATA with two FXS ports and a Wireless-G multiport IP router (see

Figure 1-11). The WRP400 provides connectivity to an analog telephone as well as Internet connectivity

to a LAN with a built-in four-port switch. The IP router provides QoS support, an integrated firewall and VPN pass-through, and full routing with DHCP and NAT.

Figure 1-11 WRP400

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The following tables describe the LEDS on the front panel and the ports on the back panel of the device

Table 1-18 WRP400 Front Panel

LED Function

Ethernet 1-4 Steady green indicates an active connection to the network. Flashing

indicates traffic.

Wireless Steady green indicates an active connection to the network. Flashing

indicates traffic.

Phone 1/2 Steady green when telephone or fax machine has an active or registered

connection to the ITSP through the Phone port. Flashing indicates phone is in use or off hook.

Internet Steady green indicates active connection. Flashing indicates traffic.

Table 1-19 WRTP54G Back Panel

Port Function

Internet Connect to the broadband (cable/DSL) modem using an Ethernet cable.

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

Ethernet 1-4 Connect to local IP devices, such as PCs, using an Ethernet cable.

Power Connect to the 12-volt power supply.

WRTP54G

The WRTP54G provides an ATA with two FXS ports and a Wireless-G multiport IP router (see

Figure 1-12). The WRTP54G provides connectivity to an analog telephone as well as Internet

connectivity to a LAN with a built-in four-port switch. The IP router provides QoS support, an integrated firewall and VPN pass-through, and full routing with DHCP and NAT.

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Figure 1-12 WRTP54G

Chapter 2 Getting Started

The following tables describe the LEDS on the front panel and the ports on the back panel of the device

Table 1-20 WRTP54G Front Panel

LED Function

Ethernet 1-4 Steady green indicates an active connection to the network. Flashing

indicates traffic.

Wireless Steady green indicates an active connection to the network. Flashing

indicates traffic.

Phone 1/2 Steady green when telephone or fax machine has an active or registered

connection to the ITSP through the Phone port. Flashing indicates phone is in use or off hook.

Internet Steady green indicates active connection. Flashing indicates traffic.

Table 1-21 WRTP54G Back Panel

Port Function

Internet Connect to the broadband (cable/DSL) modem using an Ethernet cable.

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

Ethernet 1-4 Connect to local IP devices, such as PCs, using an Ethernet cable.

Power Connect to the 12-volt power supply.

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WRT54GP2

Linksys Analog Telephone Adapters (ATAs)

The WRP54GP2 provides an ATA with two FXS ports and a Wireless-G multiport IP router (see

Figure 1-13). The WRP54GP2 provides connectivity to an analog telephone as well as Internet

connectivity to a LAN with a built-in four-port switch. The IP router provides QoS support, an integrated firewall and VPN pass-through, and full routing with DHCP and NAT.

Figure 1-13 WRT54GP2

The following tables describe the LEDS on the front panel and the ports on the back panel of the device.

Table 1-22 WRT54G P2 Front Panel

LED Function

Power Steady green indicates that the power supply is connected and the device is

functioning.

WLAN Steady green indicates an active connection to the network. Flashing

indicates traffic.

Ethernet 1-3 Steady green indicates an active connection to the network. Flashing

indicates traffic.

Phone 1/2 Steady green when telephone or fax machine has an active or registered

connection to the ITSP through the Phone port. Flashing indicates phone is in use or off hook.

Internet Steady green indicates active connection. Flashing indicates traffic.

Table 1-23 WRTP54G Back Panel

Port Function

Internet Connect to the broadband (cable/DSL) modem using an Ethernet cable.

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

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Establishing Connectivity

Table 1-23 WRTP54G Back Panel

Port Function

Ethernet 1-3 Connect to local IP devices, such as PCs, using an Ethernet cable.

Power Connect to the power supply.

Establishing Connectivity

This section describes how to connect the Linksys ATA hardware. It includes the following topics:

• Bandwidth Requirements, page 2-14

• Making the Physical Connections, page 2-15

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.

Chapter 2 Getting Started

Linksys recommends using the Linksys ATA 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. illustrates the bandwidth budget using different codecs.

Table 1-24 Ethernet Bandwidth Budget for Off-Net VoIP Calling

Approximate bandwidth budget for each

Codec

G.711 110 kbps 220 kbps 440 kbps 660 kbps 880 kbps

G.726-40 87 kbps 174 kbps 348 kbps 522 kbps 696 kbps

G.726-32 79 kbps 158 kbps 316 kbps 474 kbps 632 kbps

G.726-24 71 kbps 142 kbps 284 kbps 426 kbps 568 kbps

G.726-16 63 kbps 126 kbps 252 kbps 378 kbps 504 kbps

G.729 55 kbps 110 kbps 220 kbps 330 kbps 440 kbps

This table is based on the following assumptions:

• Bandwidth Calculated with No Silence Suppression

• 20 milliseconds of payload per RTP packet

side of conversation

2 calls 4 calls 6 calls 8 calls

Table 1-24

2-14

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:

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

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

Making the Physical Connections

Make sure that you have the following package contents:

• Linksys phone adapter unit

• Ethernet cable

• RJ-11 phone cable (SPA3102/AG310 Only)

• SPA Quickstart Guide

• Volt power adapter

You also need the following:

• One or two analog touchtone telephones (or fax Machine)

• Access to an IP network via an Ethernet connection for WAN connectivity

• Access to a PSTN network connection (SPA3102/AG310 only).

Establishing Connectivity

Perform the following steps to connect the Linksys ATA.

Step 1 Insert a standard RJ-45 Ethernet cable (included) into the LAN port and connect the other end to the

broadband router or modem.

Step 2 (Optional) For a Linksys ATA with more than one Ethernet port, you can connect the extra port to a PC

or other Ethernet device.

Step 3 Insert a standard RJ-11 telephone cable into the Phone 1 port.

Step 4 (Optional) For a Linksys ATA with more than one FXS port, you can connect the extra port to a second

analog telephone or a fax machine.

Note Do not connect an RJ-11 telephone cable from the Linksys ATA to the wall jack to prevent any

chance of connection to the circuit switched Telco network.

Step 5 (SPA3102/AG310 only) Connect the RJ-11 phone cable (included) to the SPA3102 LINE port and to

your telephofne wall jack.

Step 6 Connect the included power adapter to the Linksys ATA power port, and then plug the power adapter

into an electrical outlet.

The power LED on the front panel will light up as soon as the device powers on.

Step 7 Power on the broadband modem or router.

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Connecting the SPA8000

The SPA8000 provides up to eight analog telephone connections, and is designed to function as a network endpoint. This section describes the architecture and connectivy requirements of the SPA8000 and includes the following topics:

• SPA8000 Architecture, page 2-16

• Connectivity Requirements, page 2-17

SPA8000 Architecture

As shown in Figure 1-14, the SPA8000 consists of four hardware modules: one primary (Module 1) and three secondary modules (Modules 2, 3, and 4). Each module provides two FXS ports. The primary module supports Line 1 and Line 2, while the three secondary modules support Line 3-4, 5-6, and 7-8.

Figure 1-14 SPA8000 Internal Configuration

8 FXS (RJ-

11/RJ-21 ) ports

Chapter 2 Getting Started

Administrative IVR (Line 1 or

Line 2)

Module 1 (primary)

Line 1

Line 2

Line 3

Line 4

Line 5

Line 6

Line 7

Line 8

SPA8000

NAT/PAT

Internal DHCP

server

192.168.0.1

Module 2 (secondary)

192.168.0.2

Module 3 (secondary)

192.168.0.3

Module 4 (secondary)

192.168.0.4

192.168.1.101

Ethernet

port

AUX

port

IP Router (with hairpinning) or

Broadband modem

192.168.1.100

192.168.0.5

Secondary

Module 2

ISP

209.165.202.129

Administration web server

Internet

ITSP

The four modules are interconnected through an internal IP network, configured automatically as a single subnet, which by default is assigned the network address 192.168.0.0 with a subnet mask of

255.255.255.0. If a network conflict is detected from the device connected on the AUX port, the internal network is automatically changed to 192.168.1.0.

2-16

The primary module is responsible for the management functions of the entire SPA8000, in addition to handling calls on Line 1 and Line 2. These management functions include remote provisioning, firmware upgrades, the administration web server, IVR functions, and SIP NOTIFY control messages (over Line 1 or 2). The IVR functions are accessed by connecting an analog telephone to Line 1 or Line 2.

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Connecting the SPA8000

The secondary modules (Module 2, 3, and 4) obtain configuration and firmware upgrades from the primary module and are not managed directly. The primary module routes all the SIP and RTP traffic to and from the secondary modules over the SPA8000 internal LAN, which also includes the AUX maintenance port. The AUX maintenance port cannot be used to forward IP traffic. It is designed to connect to the management functions on the SPA8000, such as the administration web server.

The secondary modules are assigned IP addresses by the DHCP server that runs on the primary module. The primary module also performs Network Address Translation (NAT) and Port Address Translation (PAT) on the IP addresses and ports used by the secondary modules, translating them into addresses and port numbers that can be transmitted over the network connected to the Ethernet port.

For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the SPA8000 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP for each of the four modules.

When performing a firmware upgrade, either through remote provisioning or through the administration web server, the SPA8000 downloads and upgrades the primary module first. After successful upgrade of the primary module, the three secondary modules update their firmware image directly from the primary module.

The primary module is also connected through the Ethernet port to the external local area network, or directly to the Internet. This modules receives its IP address through DHCP, or its address can be statically assigned using the IVR or the administration web server. To access the IVR, connect an analog telephone to the Phone 1 or Phone 2 port. To access the administration web server, connect a PC to the AUX port.

Note In the illustration, one fax machine is connected to each pair of ports to illustrate that only one T.38

connection is supported by each of the four pairs of RJ-11 ports. Up to four fax machines can be connected to the SPA8000, but they must be distributed as shown.

The secondary modules (Line 3/4, 5/6, and 7/8) are assigned IP addresses by an internal DCHP server provided by the SPA8000, and these addresses are not configurable.

Note The SPA8000 is not designed to forward IP packets to devices connected to its AUX port and that

configuration is not supported.

Connectivity Requirements

The SPA8000 can be connected to a local router through its LAN port, or directly to the Internet. Each phone connected to an RJ-11 (analog) port on the SPA8000 connects to other devices through SIP, which is transmitted over the IP network.

In order to ensure connectivity between the devices connected to its FXS ports, the SPA8000 requires the following functionality to be supplied on the network connected to its Ethernet port:

• Connection to an IP router with hairpinning support

• Connection to an outbound Proxy server

When a phone connected to a port on the SPA8000 communicates with another phone, it sends a SIP packet onto the internal LAN, which is then forwarded to the external local area network or directly to the Internet. The source address and source port on the original packet is assigned by the SPA8000

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DHCP server, and the address and port is translated by the SPA8000 using Network Address Translation (NAT) and Port Address Translation (PAT). The packet must then be routed back to the internal network on the SPA8000 by the local router or the ISP router.

Problems can occur with calls between phones connected to the SPA8000 when an outbound proxy or a router with hairpinning support is not available. The SPA8000 cannot directly connect the two telephone devices, but requires a local or remote router to route the packet back to its destination on the local network from which it originated.

The necessary routing can be provided by a router with hairpinning support, or by an outbound SIP proxy, which is typically provided by the Internet Telephony Service Provider (ITSP). When relying on the ITSP for interconnecting phones on the SPA8000, local phones connected to the SPA8000 are unable to communicate with each other if the Internet connection is not available for any reason. It is recommended you connect the SPA8000 to a local router that provides hairpinning support to prevent this problem.

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 Linksys ATA. It includes the following topics:

• Using the IVR Menu, page 2-18

• IVR Options, page 2-19

Chapter 2 Getting Started

• Entering a Password through the IVR, page 2-21

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.

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 Linksys ATA.

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.

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Step 3 Refer to Table 1-25 to identify the option required.

Step 4 Enter the required option followed by the # (pound) key.

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Step 5 To exit the menu, hang up the telephone.

Using the Interactive Voice Response Interface

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 Linksys ATA times out. You need to re-enter the menu by pressing ****.

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

IVR Options

Table 1-25 summarizes the options provided by the IVR.

Table 1-25 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 3948 None

Check DHCP 100 None IVR announces if DHCP in enabled or

disabled.

Enable/Disable DHCP 101 Enter 1 to enable

Enter 0 to disable

Check WAN IP Address 110 None IVR announces the current IP address of

Set Static IP Address 111 Enter IP address using

numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Requires password

the WAN port.

DHCP must be “Disabled,” otherwise you hear, “Invalid Option,” if you try to set this value.

Requires password

Check Network Mask 120 None IVR announces the current network

mask of Linksys ATA.

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Table 1-25 IVR Options (continued)

Chapter 2 Getting Started

Set Network Mask 121 Enter value using numbers

on the telephone key pad. Use the * (star) key when entering a decimal point.

Check Static Gateway IP

130 None IVR announces the current gateway IP

Address

Set Static Gateway IP Address

131 Enter IP address using

numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

DHCP must be “Disabled,” otherwise you hear, “Invalid Option,” if you try to set this value.

Requires password

address of the Linksys ATA.

DHCP must be “Disable,” otherwise you hear, “Invalid Option,” if you try to set this value.

Requires password

Check MAC Address 140 None IVR announces the MAC address of

Linksys ATA in hex string format.

Check Firmware Version 150 None IVR announces the version of the

firmware running on the Linksys ATA.

Check Primary DNS Server Setting

Set Primary DNS Server 161 Enter IP address using

160 None IVR announces the current setting in the

<Primary DNS> parameter.

Requires password numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Check administration web server port

170 None IVR announces the port that the web

server is listening on. (Default is 80)

Check LAN IP Address 210 None IVR announces the current IP address of

the LAN port.

Check PBX multicast

180 None IVR announces the current value.

address

Set PBX multicast address

181 Enter IP address and port.

Use * key for entering a dot.

Enter a * between the IP address and the

Port fields. Requires Password For example,

224.168.168.169:8089 is 224*168*168*169*8089.

Enable/Disable administration web

7932 Enter 1 to enable

Enter 0 to disable

Requires password

server

Manual Reboot of Unit 732668 None After you hear “Option Successful,”

hang up. Unit reboots automatically.

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Table 1-25 IVR Options (continued)

Using the Interactive Voice Response Interface

User Factory Reset of Unit

WARNING:

ALL “User-Changeable” NON-DEFAULT SETTINGS WILL BE LOST!

This might include network and service provider data.

Factory Reset of Unit

WARNING:

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.

877778 Enter 1 to confirm

Enter *(star) to cancel operation

73738 (RESET) Enter 1 to confirm

Enter * (star) to cancel operation

Linksys ATA 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.

Linksys ATA prompts for confirmation. After confirming, you hear “Option Successful.” Hang up. Unit reboots and all configuration parameters are reset to factory default values.

Entering a Password through the IVR

To input the password using the phone keypad, the following translation convention applies:

–

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.

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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.

–

Notes:

–

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

from the beginning by pressing * * * *.

Chapter 2 Getting Started

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).

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

Saved settings take effect when the telephone is hung-up, and if necessary, the Linksys ATA automatically reboots.

Using the Administration Web Server

This section describes how to use the administration web server to configure the Linksys ATA. It includes the following topics:

• Connecting to the Administration Web Server, page 2-22

• Administrator Account Privileges, page 2-23

Connecting to the Administration Web Server

To access the Linksys ATA administration web server, perform the following steps.

Step 1 Launch a web browser on a computer connected to the Linksys ATA.

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

Step 2 Determine the address of the administration web server.

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

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

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

Step 3 Direct the browser to the IP address of the Linksys ATA.

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.

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

Step 4 Click the Voice tab.

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Step 5 Click Admin and Advanced.

The Administrator account name is admin, 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.

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 Linksys ATA may reboot.

Administrator Account Privileges

The Linksys ATA supports two levels of administration privileges: Administrator and User. Both privileges can be password protected.

Using the Administration Web Server

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 using the Administrator account on the Provisioning page of the administration web server.

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

http://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

Login link). From the User account, you can switch to the Administrator account by clicking the Admin Login link. Authentication is required if the Administrator account password has been set.

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.

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CHA PTER

Configuring Linksys ATAs

This chapter describes how to perform site-specific configuration required to use a Linksys ATA or to enable specific features. It includes the following sections:

• Initial Configuration, page 3-1

• Web Interface URLs, page 3-3

• Provisioning, page 3-4

• Configuring a Dial Plan, page 3-5

• Secure Call Implementation, page 3-10

• Configuring a Streaming Audio Server, page 3-12

• Using a FAX Machine with the SPA2102 or SPA8000, page 3-15

• Managing Caller ID Service, page 3-17

• Troubleshooting and Configuration FAQ, page 3-18

Initial Configuration

Step 1 After connecting the Linksys ATA to your LAN, launch the web browser on your PC and direct it to the

following URL:

192.168.0.1/advanced

192.168.0.1 is the default local IP address of a Linksys ATA that provides a built-in IP router.

If the ITSP supplied a password for administrative access, you will see a login screen. In the User Name field, enter user, the default user name for user access, or admin for administrator access. Then enter the user password supplied by your ITSP.

If your ATA includes a built-in IP router, the Router - Status screen appears.

Step 2 Click the WAN Setup tab.

Step 3 Complete the WAN configuration for DHCP, static IP addressing, or PPPoE.

If you are using DHCP for dynamically assigning the IP address to the Linksys ATA, complete the following steps:

a. Select DHCP from the Connection Type drop-down menu.

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If you use a cable modem, you may need to configure the MAC Clone Settings. (Contact your ISP for more information.)

b. If your service uses a specific PC MAC address, then select yes from the Enable MAC Clone Service

setting.

c. Then enter the PC’s MAC address in the Cloned MAC Address field.

If you are using static IP addressing, complete the following steps:

a. Select Static IP from the Connection Type drop-down menu.

b. In the Static IP Settings section, enter the IP address in the Static IP field, the subnet mask in the

NetMask field, and the default gateway IP address in the Gateway field.

c. In the Optional Settings section, enter the DNS server address(es) in the Primary DNS and optional

Secondary DNS fields.

If you are using PPPoE, complete the following steps:

a. Select PPPoE from the Connection Type drop-down menu.

This is the correct setting for most DSL users.

b. Enter the values provided by the ITSP in the following fields:

• PPPoE Login Name

• PPPoE Login Password

• PPPoE Service Name

Step 4 Configure the Linksys ATA parameters on the Line tab with the appropriate values for the specific ITSP.

If the Linksys ATA has more than one Line tab, each line tab must be configured separately. Each line tab can be configured for a different ITSP.

The minimum required configuration to connect the Linksys ATA to an ITSP includes configuring the following parameters on the appropriate Line tab with the values provided for a specific ITSP:

• User ID (Subscriber Information section)

• Password (Subscriber Information section)

• Proxy (Proxy and Registration section)

Other parameters may need to be changed from the defaults, depending on the requirements of a specific ITSP. Some of the commonly changed sections include the following:

• Streaming Audio Server—to enable an external music source for enabling music on hold (see the

“Configuring a Streaming Audio Server” section on page 3-12 for further information).

• NAT Settings—to workaround NAT issues (see the “Troubleshooting and Configuration FAQ”

section on page 3-18 for further information)

• Subscriber Information—to configure security parameters (see the “Secure Call Implementation”

section on page 3-10 for further information)

• Dial Plan—to configure a dial plan for a specific line (see the “Configuring a Dial Plan” section on

page 3-5 for further information)

Step 5 After making any necessary changes to the Linksys ATA parameters, click the Submit All Changes

button.

The changes are saved, the Linksys ATA reboots, and the configuration changes are applied to the running configuration.

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Web Interface URLs

The Linksys ATA web interface supports several functions through special URLs:

• Upgrade

• Reboot

• Resync

Administrator account privilege is needed for these functions.

Upgrade URL

The Upgrade URL lets you upgrade the Linksys ATA 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

Linksys ATA even if the web page indicates otherwise.

Web Interface URLs

Resync URL

Note The SPA resyncs only when it is idle.

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

The Resync URL lets you force the Linksys ATA to do a resync to a profile specified in the URL, which can identify either a TFTP, HTTP, or HTTPS server.

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

is specified, /spa.bin is assumed, as in the following example:

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Provisioning

Reboot URL

The Reboot URL lets you reboot the Linksys ATA.

Note The Linksys ATA 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 Linksys ATA. This section includes the following topics:

• Provisioning Capabilities, page 3-4

• Configuration Profile, page 3-4

For detailed information about provisioning the Linksys ATA, refer to the Linksys SPA Provisioning Guide.

Chapter 3 Configuring Linksys ATAs

Provisioning Capabilities

The Linksys ATA provides for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles transferred to the device via TFTP, HTTP, or HTTPS.

The Linksys ATA 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 Linksys ATA accepts profiles in XML format, or alternatively in a proprietary binary format, which is generated by a profile compiler tool available from Linksys. The Linksys ATA supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial provisioning stage), the Linksys ATA 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 Linksys ATA, 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 Linksys ATA via a remote profile resync.

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

Configuration Profile

The Linksys ATA 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>

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<Upgrade_Enable>Yes</Upgrade_Enable> </flat-profile>

Binary format profiles contain Linksys ATA parameter values and user access permissions for the parameters. By convention, the profile uses the extension .cfg (for example, spa2102.cfg). The Linksys Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly 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 Linksys SPA Provisioning Guide for further details.

The names of parameters in XML profiles can generally be inferred from the Linksys ATA 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.

Configuring a Dial Plan

The Linksys ATA 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 Linksys ATA 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.

Note When using the SPA3102 or AG310 as a PSTN gateway, gateway calls can be restricted on a per-caller

basis using dial plans. Up to eight dial plans can be configured to restrict gateway calls in either direction.

Dial Plan Digit Sequences

The plans contain a series of digit sequences, separated by a vertical bar ( | ). 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 Linksys ATA 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.

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• 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.

The following table describes the entries to use when programming the dial plan.

Table 1-26 Dial Plan Entries

Dial Plan Entry Function

*xx Allows arbitrary 2-digit star code

[3469]11 Allows x11 sequences (for example, 311, 411, 611, 911)

0 Dials operator

00 Dials international operator

[2-9]xxxxxx Dials US local number

1xxx[2-9]xxxxxx Dials US 1 + 10-digit long distance number

xxxxxxxxxx. Dials all other numbers, including international long distance

Note Early production versions of the SPA2102 supported dual-line telephones on a single FXS port. A

subsequent hardware change revised the FXS ports on the SPA2102 and this function is to be reserved for future development.

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)

–

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–

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

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.

Configuring a Dial Plan

Number Barring

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.

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

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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 Linksys ATA 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 )

Chapter 3 Configuring Linksys ATAs

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. )

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 )

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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 Interdigit_Long_Timer

Default 10

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).

Configuring a Dial Plan

Interdigit Short Timer

ParName Interdigit_Short_Timer

Default 3

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 Dial_Plan[n] for Each Line n

Default

The Dial_Plan parameters contain the actual dial plan scripts for each line n, where n is a number from 1 to 4.

(*xx | [3469]11 | 0 | 00 | <:1408>[2-9]xxxxxx |

1[2-9]xx[2-9]xxxxxx | 011x. )

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Secure Call Implementation

This section describes secure call implementation with a Linksys ATA. It includes the following topics:

• Enabling Secure Calls, page 3-10

• Secure Call Details, page 3-10

• Using a Mini-Certificate, page 3-11

• Generating a Mini-Certificate, page 3-11

Enabling Secure Calls

A secure call is established in two stages. The first stage is no different from normal call setup. The second stage starts after the call is established in the normal way with both sides ready to stream RTP packets. I

In the second stage, the two parties exchange information to determine if the current call can switch over to the secure mode. The information is transported by base64 encoding embedded in the message body of SIP INFO requests, and responses using a proprietary format. If the second stage is successful, the Linksys ATA plays a special Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted.

If the user has a phone that supports call waiting caller ID (CIDCW) and that service is enabled, the CID will be updated with the information extracted from the Mini-Certificate received from the remote party. The Name field of the CID will be prepended with a ‘$’ symbol. Both parties can verify the name and number to ensure the identity of the remote party.

The signing agent is implicit and must be the same for all Linksys ATAs that communicate securely with each other. The public key of the signing agent is pre-configured into the Linksys ATA by the administrator and is used by the Linksys ATA to verify the Mini-Certificate of its peer. The Mini-Certificate is valid if it has not expired, and it has a valid signature.

The Linksys ATA can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If not secure by default, the user can make a secure outbound call by dialing *18 before dialing the target number. However, the user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

The Linksys ATA will not switch to secure mode if the CID of the called party from its Mini-Certificate does not agree with the user-id used in making the outbound call. The Linksys ATA performs this check after receiving the Mini-Certificate of the called party

Chapter 3 Configuring Linksys ATAs

Secure Call Details

Looking at the second stage of setting up a secure call in greater detail, this stage can be further divided into two steps.

1. The caller sends a “Caller Hello” message (base64 encoded and embedded in the message body of

a SIP INFO request) to the called party with the following information:

• Message ID (4B)

• Version and flags (4B)

• SSRC of the encrypted stream (4B)

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• Mini-Certificate (252B)

Upon receiving the Caller Hello, the called party responds with a Callee Hello message (base64 encoded and embedded in the message body of a SIP response to the caller’s INFO request) with similar information, if the Caller Hello message is valid. The caller then examines the Callee Hello and proceeds to the next step if the message is valid.

2. The caller sends the “Caller Final” message to the called party with the following information:

• Message ID (4B)

• Encrypted Master Key (16B or 128b)

• Encrypted Master Salt (16B or 128b)

The Master Key and Master Salt are encrypted with the public key from the called party mini-certificate. The Master Key and Master Salt are used by both ends for deriving session keys to encrypt subsequent RTP packets. The called party then responds with a Callee Final message (which is an empty message).

Using a Mini-Certificate

The Linksys ATA Mini-Certificate (MC) contains the following information:

• User Name (32B)

• User ID or Phone Number (16B)

• Expiration Date (12B)

Secure Call Implementation

• Public Key (512b or 64B)

• Signature (1024b or 512B)

The MC has a 512-bit public key used for establishing secure calls. The administrator must provision each subscriber of the secure call service with an MC and the corresponding 512-bit private key. The MC is signed with a 1024-bit private key of the service provider, which acts as the CA of the MC. The 1024-bit public key of the CA signing the MC must also be provisioned for each subscriber.

The CA public key is used by the Linksys ATA to verify the MC received from the other end. If the MC is invalid, the Linksys ATA will not switch to secure mode. The MC and the 1024-bit CA public key are concatenated and base64 encoded into the single parameter <Mini Certificate>. The 512-bit private key is base64 encoded into the <SRTP Private Key> parameter, which should be kept secret, like a password.

Because the secure call establishment relies on exchange of information embedded in message bodies of SIP INFO requests/responses, the service provider must ensure that the network infrastructure allows the SIP INFO messages to pass through with the message body unmodified.

Generating a Mini-Certificate

Linksys provides a configuration tool called gen_mc for the generation of MC and private keys with the following syntax:

gen_mc ca-key user-name user-id expire-date

Where:

• ca-key is a text file with the base64 encoded 1024-bit CA private/public key pairs for

signing/verifying the MC, such as the following:

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9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYx WCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTj j13qvYs=

5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/IqSrsf6scsmundY5j7Z5mK 5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3MF+zjyYrVUFdM+pXtDBxmM+fGUfrp AuXb7/k=

• user-name is the name of the subscriber, such as “Joe Smith”. Maximum length is 32 characters

• user-id is the User ID of the subscriber, which must match exactly the user-id used in the INVITE

when making the call, such as “14083331234”. The maximum length is 16 characters.

• expire-date is the expiration date of the MC, such as “00:00:00 1/1/34” (34=2034). Internally the

date is encoded as a fixed 12B string: 000000010134

The tool generates the <Mini Certificate> and <SRTP Private Key> parameters that can be provisioned to the Linksys ATA.

For Example:

gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34”

Produces the following Mini Certificate and SRTP Private Key:

<Mini Certificate> Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEwMTM00OvJakde2v VMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L 4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhxES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2Yd mTMBR516xJt4/uQ/LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3Amc qE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1 fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs= <SRTP Private Key> b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ==

Chapter 3 Configuring Linksys ATAs

Configuring a Streaming Audio Server

This section describes how to use and configure a streaming audio server (SAS). It includes the following topics:

• Music On Hold, page 3-12

• Using a Streaming Audio Server, page 3-13

• Using the IVR with an SAS Line, page 3-13

• Example SAS with MOH, page 3-14

• SAS Line Not Registered with the Proxy Server, page 3-14

Music On Hold

On a connected call, the Linksys ATA 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 Linksys ATA 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.

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Using a Streaming Audio Server

The SAS feature lets you use attach an audio source to one of the Linksys ATA FXS ports (Phone 1 or Phone 2 on the PAP2T) and use it as a streaming audio source device. If the Linksys ATA 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

To connect an external music source to an FXS port, use a media signal adapter, which provides a line in from a media source and a RJ-11 port for connecting to the FXS port on the Linksys ATA. The following is a URL for a device that has been tested with Linksys ATAs:

http://www.neogadgets.com/cart/cart.php?target=product&product_id=17&substring=music+coupler

After installing the music source using the media signal adapter and completing the required configuration on the Linksys ATA , when the line is called and the FXS port is off hook, the Linksys ATA answers the call automatically and streams audio to the caller.

If the FXS port is on-hook when the incoming call arrives, the Linksys ATA replies with a SIP 503 response code (Service Not Available). The SAS line will not ring for incoming calls even if the attached equipment is on-hook.

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 (Busy Here).

The SAS line can be set up 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 Linksys ATA is disabled, then the SAS line can maintain up to 10 simultaneous calls. Further incoming calls will receive a busy signal (SIP 486 Response).

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.

Configuring a Streaming Audio Server

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:

Step 1 Power off the Linksys ATA.

Step 2 Connect a phone to the port and make sure the phone is on-hook.

Step 3 Power on the Linksys ATA.

Step 4 Pick up handset and press * * * * to invoke IVR in the usual way.

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If the Linksys ATA 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 Linksys ATA 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 1-15 MOH Application with a Linksys ATA Line Configured as an SAS

Chapter 3 Configuring Linksys ATAs

SPA1: IP=192.168.2.100 UserID[1]=1001, SIP Port[1]=5060 UserID[1]=1002, SIP Port[1]=5061

Phone 1

Phone 2

Media signal adapter

Music source

The following configuration examples are based on the configuration shown in Figure 1-15.

SAS Line Registered with the Proxy Server

In this example, the SAS Line is registered with the Proxy Server as the other subscribers.

On Linksys ATA 1:

SAS Enable[1] = no MOH Server [1] = 1002 SAS Enable[2] = yes

SPA2: IP=192.168.2.200 UserID[1]=2001, SIP Port[1]=5060 UserID[1]=2002, SIP Port[1]=5061

IP Network

Phone 1

Phone 2

Line in

On Linksys ATA 2:

SAS Enable[1] = no MOH Server [1] = 1002 SAS Enable[2] = no MOH Server [2] = 1002

SAS Line Not Registered with the Proxy Server

In this example, the SAS Line is not registered with the Proxy Server for the other subscribers.

On Linksys ATA 1:

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SAS Enable[1] = no MOH Server [1] = 1002@192.168.2.100:5061 or 1002@127.0.0.1:5061 SAS Enable[2] = yes

On Linksys ATA 2:

SAS Enable[1] = no MOH Server [1] = 1002@192.168.2.100:5061 SAS Enable[2] = no MOH Server [2] = 1002@192.168.2.100:5061

Configuring the Streaming Audio Server

The following provides step-by-step procedures for implementing an SAS with an external music source.

Step 1 Connect an RJ-11 adapter between the music source and an FXS port on the Linksys ATA (Phone 1 or

Phone 2).

Step 2 On the administration web server, click the SIP tab and scroll down to the Streaming Audio Server

section (see

Figure 1-16).

Using a FAX Machine with the SPA2102 or SPA8000

Figure 1-16 Enabling the Streaming Audio Server

Step 3 On the SAS Enable pull-down selection list, select yes.

Step 4 In the MOH Server field of the Call Feature Settings section, enter the User ID configured for the line

attached to the audio source.

If the line is not registered with the SIP proxy, enter the IP address and SIP port number configured for the line.

Step 5 Click Submit All Changes.

Using a FAX Machine with the SPA2102 or SPA8000

Note T.38 Fax is only supported on the SPA2102 and the SPA8000. The SPA2102 supports a single

connection, whilethe SPA8000 supports one connection for each pair of ports (1/2, 3/4, 5/6, and 7/8) for a maximum of four connections.

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Procedure

To optimize fax completion rates, complete the following steps:

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Using a FAX Machine with the SPA2102 or SPA8000

Step 1 Upgrade the ATA firmware to the latest version

Step 2 Ensure that you have enough bandwidth for uplink and downlink.

• For G.711 fallback, it is recommend to have ~100Kbps.

• For T.38, allocate at least 50 kbps.

Step 3 To optimize G.711 fallback fax completion rates, use the following settings on the Line tab of the

Linksys ATA:

• Network Jitter Buffer: very high

• Jitter buffer adjustment: disable

• Call Waiting: no

• 3 Way Calling: no

• Echo Canceller: no

• Silence suppression: no

• Preferred Codec: G.711

• Use pref. codec only: yes

Step 4 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax

using modem passthrough.

Chapter 3 Configuring Linksys ATAs

For example:

modem passthrough nse payload-type 110 codec g711ulaw fax rate disable fax protocol pass-through g711ulaw

Step 5 Enable T.38 fax on the SPA 2102 by configuring the following parameter on the Line tab for the FXS

port to which the FAX machine is connected:

FAX_Passthru_Method: ReINVITE

Note If a T.38 call cannot be set-up, then the call should automatically revert to G.711 fallback.

Step 6 If you are using a Cisco media gateway use the following settings:

Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial peer. For example:

fax protocol T38 fax rate voice fax-relay ecm disable fax nsf 000000 no vad

Troubleshooting

If have problems sending or receiving faxes, complete the following steps:

3-16

Step 1 Send a test Fax in a controlled environment between two Linksys ATAs.

Step 2 Determine the success rate.

Step 3 Monitor the network and record the following statistics:

• Jitter

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• Loss

• Delay

Step 4 If faxes fail consistently, capture a copy of the web interface settings by selecting Save As > Web page,

complete from the administration web server page.

Step 5 Enable and capture the debug log. For instructions, refer to Item 4. in “Troubleshooting and

Configuration FAQ” at the end of this chapter.

Step 6 Identify the type of Fax machine connected to the ATA.

Step 7 Contact technical support.

If you are an end user of Linksys VoIP products, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment.

If you are an authorized Linksys Voice System partner, contact Linksys technical support.

Managing Caller ID Service

The choice of caller ID (CID) method affects the following features:

• On Hook Caller ID Associated with Ringing – This type of Caller ID is used for incoming calls when

the attached phone is on hook. See figure below (a) – (c). All CID methods can be applied for this type of CID.

• On Hook Caller ID Not Associated with Ringing – This feature is used for send VMWI signal to the

phone to turn the message waiting light on and off (see Figure 1 (d) and (e)). This is available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI FSK With PR).

• Off Hook Caller ID – This is used to delivery caller-id on incoming calls when the attached phone

is off hook (see

Figure 1-17). This can be call waiting caller ID (CIDCW) or to notify the user that

the far end party identity has changed or updated (such as due to a call transfer). This is available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI FSK With PR).

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Figure 1-17 Linksys ATA Caller ID Delivery Architecture

a) Bellcore/ETSI Onhook Post-Ring FSK

First Ring

b) ETSI Onhook Post-Ring DTMF

First Ring

c) ETSI Onhook Pre-Ring FSK/DTMF

d) Bellcore Onhook FSK w/o Ring

e) ETSI Onhook FSK w/o Ring

f) Bellcore/ETSI Offhook FSK

Polarity

Reversal

OSI FSK

Polarity

Reversal

CAS

(DTAS)

CAS

(DTAS)

CAS

(DTAS)

Wait For

ACK

FSK

DTMF

DTMF/

FSK

Chapter 3 Configuring Linksys ATAs

First Ring

Troubleshooting and Configuration FAQ

This section provides solutions to problems that may occur during the installation and operation of the Linksys ATA. If you can't find an answer here, check the Linksys website at www.linksys.com.

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

but this address did not work.

Use the Interactive Voice Response Menu to find out the Linksys ATA Internet IP address. Follow these steps: A. Use a telephone connected to the Phone 1 port of the Linksys ATA. B. Press **** (in other words, press the star key four times). 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 Linksys ATA Internet (external) interface. Write it down. F. Press 7932#. G. Press 1 to enable WAN access to the administration web server. H. Open the web browser on a networked computer. I. Enter http://(Internet IP address of the Linksys ATA).

2. I’m trying to access the Linksys ATA 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.

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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.

3. 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.

4. 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 Linksys ATA).

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)

Troubleshooting and Configuration FAQ

5. How do I access the Linksys ATA if I forget my password?

By default, the User and Admin accounts have no password. If the ITSP set the password for either account and you do not know what it is, you need to contact the ITSP. If the password for the user account was configured after you received the Linksys ATA, you can reset the device to the user factory default, which preserves any provisioning completed by the ITSP. If the Admin account needs to be reset, you have to perform a full factory reset, which also erases any provisioning.

To reset the Linksys ATA to the factory defaults, perform the following steps:

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

B. Press the appropriate code to reset the unit:

• Press 877778# to reset the unit to the defaults as it shipped from the ITSP. This will reset the

User account password to the default of blank.

• Press 73738# to perform a full reset of unit to the defaults as it shipped from Linksys. This will

reset the Admin account password to the default of blank.

C. Press 1 to confirm the operation.

Press * to cancel the operation.

D. Login to the unit using the User or Admin account without a password and reconfigure the unit as

necessary.

6. My Linksys ATA 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.

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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 Linksys ATA is registering, if it supports NAT, using the <Outbound

Proxy> parameter. D. 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.

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CHA PTER

Configuring the PSTN Gateway (FXO)

This chapter describes how to configure the PSTN gateway provided by Analog Telephone Adapters (ATAs) with one or more FXO ports, which includes the AG310 and SPA3102. It includes the following sections:

• Overview, page 4-1

• How VoIP-To-PSTN Calls Work, page 4-2

• How PSTN-To-VoIP Calls Work, page 4-4

• Configuring VoIP Failover to PSTN, page 4-7

• Sharing One VoIP Account Between the FXS and PSTN Lines, page 4-7

• Other Options, page 4-7

• Call Scenarios, page 4-8

Note Througout this chapter, when references are made to the software configuration of the SPA3102, the

Overview

information also applies to the AG310. The AG310 provides the same functionality as the SPA3102, as well as an ADSL modem and a four-port Ethernet switch.

The SPA3102 and AG310 have the following ports for connection to telephony devices:

• FXS port (Phone)—Connected to a standard analog telephone or fax machine, configured using the

Line tab

• FXO port (Line)—Connected to a standard telephone wall jack for connectivity to the PSTN,

configured using the PSTN Line tab

Line 1 does not provide a gateway because it provides only VoIP service. The VoIP-To-PSTN calling function is referred to as a PSTN gateway, and PSTN-To-VoIP calling function as a VoIP gateway. Note the following definitions:

• VoIP caller—One who calls the AG310 via VoIP to obtain PSTN service

• VoIP user—VoIP caller that has a user account (user-id and password) on the AG310

• PSTN caller—One who calls the AG310 from the PSTN to obtain VoIP service

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Line 1 can be configured with a regular VoIP account and can be used in the same way as the Line 1 of any Linksys ATA.

With the SPA3102 and AG310, a second VoIP account can be configured to support PSTN gateway calls exclusively. A different SIP port should be assigned to Line 1 and the PSTN Line. The same VoIP account may be used for both Line 1 and the PSTN Line if a different SIP port is assigned to each.

VoIP callers can be authenticated by one of the following methods:

• No Authentication—All callers are accepted for service

• PIN—Caller is prompted to enter a PIN right after the call is answered

• HTTP digest—SIP INVITE must contain a valid authorization header

PSTN callers can be authenticated by one of the following methods:

• No authentication—All callers are accepted for service

• PIN—Caller is prompted to enter a PIN right after the call is answered

How VoIP-To-PSTN Calls Work

Chapter 4 Configuring the PSTN Gateway (FXO)

To obtain PSTN services through the SPA3102, the VoIP caller establishes a connection with the PSTN Line by way of a standard SIP INVITE request addressed to the PSTN Line. The PSTN Line can be configured to support one-stage and two-stage dialing as described in the following sections.

One-Stage Dialing

The Request-URI of the INVITE to the PSTN Line should have the form <Dialed-Number>@<SPA-Address>, where <Dialed-Number> is the number dialed by the VoIP caller, and <SPA-Address> is a valid address of the SPA3102, such as 10.0.0.100:5061.

If the FXO port is currently in use (off-hook) or the PSTN line is being used by another extension, the SPA3102 replies to the INVITE with a 503 response. Otherwise, it compares the <Dialed-Number> with the <User ID> of the PSTN Line. If they are the same, the SPA3102 interprets this as a request for two-stage dialing (see the processes the <Dialed-Number> using the corresponding <Dial Plan>.

If dial plan processing fails, the SPA3102 replies with a 403 response. Otherwise, it replies with a 200 and at the same time takes the FXO port off hook and dials the target number returned after processing the dial plan.

Note If <User ID> on the PSTN Line is blank, <Registration> should be disabled for the PSTN Line.

If HTTP Digest Authentication is enabled, the SPA3102 challenges the INVITE with a 401 response if it does not have a valid Authorization header. The Authorization header should include a <User ID n> parameter, where n refers to one of eight VoIP user accounts that can be configured on the SPA3102. The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The <User ID n> must match one of the VoIP accounts stored on the SPA3102. Each VoIP user account contains the information listed in

“Two-Stage Dialing” section on page 4-3). If they are different, the SPA3102

Table 1-27.

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Table 1-27 VoIP User Account Information

Parameter Group Description Values

User ID 1/2/3/4/5/6/7/ 8

Password 1/2/3/4/5/6/7/ 8

User 1/2/3/4/5/6/7/ 8 DP

Note If Authentication is disabled, a default dial plan is used for all unknown VoIP users.

PSTN Line

How VoIP-To-PSTN Calls Work

The username value. 31-character string

The password value. 31-character string

Specifies the dial plan to be used for this VoIP user. If 0, dial plan processing is disabled; the given target number is dialed to the PSTN as is.

Choice of 0-8

Two-Stage Dialing

In two-stage dialing, the SPA3102 takes the FXO port off-hook but does not automatically dial any digits after accepting the call. To invoke two-stage dialing, the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI or with a user-id that matches exactly the <User ID n> of the PSTN Line. A different user-id in the Request-URI is treated as a request for one-stage dialing if one-stage dialing is enabled, or dropped by the SPA3102 (as if no user-id is given) if one-stage dialing is disabled.

Note If Authentication is disabled, a default dial plan is assigned to all VoIP callers.

HTTP Digest Authentication can be also used for two-stage dialing, as in one-stage dialing. If using HTTP Digest Authentication or Authentication is disabled, the VoIP caller should hear the PSTN dial tone right after the call is answered (by a SIP 200 response).

If PIN Authentication is enabled, the VoIP caller is prompted to enter a PIN number after the SPA3102 answers the call. The PIN number must end with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight VoIP caller PIN numbers can be configured on the SPA3102. A dial plan can be selected for each PIN number. If the caller enters a wrong PIN or the SPA3102 times out waiting for more PIN digits, the SPA3102 tears down the call immediately with a BYE request.

Note When the source address of the INVITE is 127.0.0.1, authentication is automatically disabled because

this is a call by the local user. This applies to both one-stage and two-stage dialing.

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Table 1-28 lists the parameters used in two-stage dialing.

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Table 1-28 Two-Stage Dialing

Parameter Group Description Values

VoIP Caller 1/2/3/4/5/6/7/8 PIN

VoIP Caller 1/2/3/4/5/6/7/8 DP

PSTN Line

The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8. 31-character string

Specifies which dial plan to be used for this VoIP caller. If 0, dial plan processing is disabled; the given target number is dialed to the PSTN as is.

How PSTN-To-VoIP Calls Work

PSTN-To-VoIP calls can be made with two-stage dialing only. The only authentication method available is the PIN method.

The SPA3102 takes the FXO port off hook after a configurable number of rings. If PIN Authentication is enabled, it prompts the caller to enter the PIN number followed by a # key. The Inter-PIN-digit timeout is set at 10 seconds. Up to eight PSTN PIN numbers can be configured in the SPA3102. If the given PIN does not match any of the PSTN PIN values, the SPA3102 plays the reorder tone to the FXO port for up to 10 seconds, and then takes the FXO port on-hook. If the given PIN matches one of PSTN PIN values, the SPA3102 plays dial tone to the FXO port and is ready to accept digits for the target VoIP number from the PSTN caller. The collected digits are processed by the dial plan associated with the PIN number.

Chapter 4 Configuring the PSTN Gateway (FXO)

Choice of 1 to 8

Note If Authentication is disabled, a default dial plan is used for all PSTN callers.

Terminating Gateway Calls

There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call is terminated when either call leg is ended. The SPA3102 takes the FXO port on-hook when the call terminates so the PSTN line can be used again. The SPA3102 detects that the PSTN call leg is ended when one of the following conditions occurs during a call:

• The PSTN Line voltage drops to a very low value (this occurs if the line is disconnected from the

PSTN service or if the PSTN switch provides a CPC signal)

• A polarity reversal or disconnect tone is detected at the FXO port

• When there is no voice activity for a configurable period of time in either direction at the FXO port

When any of the above occurs, the SPA3102 takes the FXO port on hook and sends a BYE request to end the VoIP call leg. On the other hand, when the SPA3102 receives a SIP BYE from the VoIP during a call, it takes the FXO port on hook to end the PSTN call leg.

In addition, the SPA3102 can also send a refresh signal periodically to the VoIP call leg to determine whether the call leg is still up. If a refresh operation fails, the SPA3102 ends both call legs.

Table 1-29 lists parameters for terminating gateway call parameters.

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Table 1-29 Terminating Gateway Call Parameters

Parameter Group Description Values

Detect CPC: PSTN

Detect Long Silence: PSTN

Long Silence Duration: PSTN

Disconnect Tone: PSTN

Detect Polarity Reversal:

Line

PSTN Line

How PSTN-To-VoIP Calls Work

If yes, SPA3102 detects CPC as a disconnect signal. Default = Yes

If yes, SPA3102 detects prolonged silence period as a disconnect signal. Default = Yes

The minimum duration of continuous silence before the SPA3102 disconnects the call, if <Detect Long Silence> is enabled. Default = 30 (s)

Tone Script of the disconnect tone to detect. Default = “480@-30,620@-30;4(.25/.25/1+2)”. Note:

The SPA3102 supports two frequency components. If the tone has only one frequency, use the same value for both frequencies.

Each cadence segment must have the same frequency.

The level value is the threshold to detect each tone.

The total duration is the minimum duration of the tone to be recognized as the disconnect tone

If yes, SPA3102 interprets polarity reversal as a disconnect signal.

Yes or No

10-255

ToneScript

Yes or No

Detect Disconnect Tone:

Silence Threshold: PSTN

VoIP Outbound Call Routing

Calls made from Line 1 are routed through the configured Line 1 service provider, by default. You can override this behavior by IP dialing, through which the calls can be routed to any IP address entered by the user. SPA3102 allows flexible call routing with four sets of gateway parameters and configurable dial plans.

Table 1-30 lists VoIP outbound call routing parameters.

PSTN Line

Line

On an inbound PSTN call, SPA3102 disconnects on the first polarity reversal. On an outbound PSTN call, SPA3102 disconnects on the second polarity reversal (because the first polarity reversal indicates the outbound call is connected). Default = Yes

If yes, SPA3102 interprets the disconnect tone as specified in <Disconnect Tone> as the disconnect signal. Default = Yes

This is the signal energy threshold. Below this threshold is considered silence. Default = medium

Yes or No

very low, low, medium, high, very high

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Table 1-30 VoIP Outbound Call Routing Parameters

Parameters Group Description Values

Gateway 1 Line 1 Fully qualified domain name (or IP address) of a

GW1 Nat

Mapping Enable

GW1 User IDLine 1 The authentication user name when using Gateway 1.

GW2 Password

Similar for GW 2, 3, and 4

Chapter 4 Configuring the PSTN Gateway (FXO)

gateway. If the port number is not specified, 5060 is assumed. Default value is [blank]

Line 1 Whether to enable NAT mapping when using

Gateway 1. Default is “no”.

Default is [blank]

Line 1 The authentication password when using Gateway 1.

Default is [blank]

Line 1 Similar for GW 2, 3, and 4

Domain name or IP address

Yes or No

31-character string

Gateways 1 to 4 can be specified in a dial plan with the special identifier gw1, gw2, gw3, or gw4. Also, gw0 represents the internal PSTN gateway via the FXO port. You can specify in the dial plan to use gwx (x = 0,1,2,3,4) when making certain calls. In general, you can specify any gateway address in the dial plan. In addition, three parameters are added that can be used with call routing:

• usr—User-id used for authentication with the given gateway

• pwd—Password used for authentication with the given gateway

• nat—Enable or disable NAT mapping when calling the gateway

Table 1-31 lists some examples.

Table 1-31 Specifying Gateway Addresses

Example Description

<9,:>xx.<:@gw1 Dial 9 to start outside dial tone, followed by one or more

digits, and route the call to Gateway 1.

[93]11<:@gw0> Route 911 and 311 calls to the local PSTN gateway

<8,:1408>xxxxxxx<:@pstn.Linksys.c om:5061;usr=joe;pwd=joe_pwd;nat>

Dial 8 to start outside dial tone, prepend 1408 followed by seven digits, and route the call to pstn.Linksys.com:5061, with user-id = joe, and pwd = bell_pwd, and enable NAT mapping

<8,:1408>xxxxxxx<:@gw2:5061;usr =”Alex Bell”;pwd=”anything”;nat=no>

Dial 8 to start outside dial tone, prepend 1408 followed by seven digits, and route the call to Gateway 2, but use the given port, user-id, and password, and no pstn.Linksys.com:5061, and with user-id = “Alex Bell” and pwd = bell_pwd, and disable NAT mapping

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You can set up multiple PSTN gateways at different locations and configure Line 1 to use a different gateway when dialing specific numbers.

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Configuring VoIP Failover to PSTN

When power is disconnected from the SPA3102, the FXS port is connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service via the FXO port. When power is applied to the SPA3102, the FXS port is disconnected from the FXO port. However, if the PSTN line is in use when the power is applied to the SPA3102, the relay is not flipped until the PSTN line is released. This is done so that the SPA3102 does not interrupt any call in progress on the PSTN line.

When Line 1 VoIP service is down (because of registration failure or loss of network link), SPA3102 can be configured to automatically route all outbound calls to the internal gateway using the parameters listed in

Table 1-32 Automatically Routing Outbound Calls to Internal Gateway

Parameter Group Description Range

Auto PSTN Fallback Line 1 If enabled, SPA3102 automatically routes

Table 1-32.

outbound calls to Gateway 0 when registration fails or network link is down.

Configuring VoIP Failover to PSTN

Bool

Sharing One VoIP Account Between the FXS and PSTN Lines

The FXS (Line 1) and FXO (PSTN Line) can share a single VoIP account if they use different SIP ports. If the service provider allows multiple registration contacts and simultaneous ringing, both lines can register periodically with the service provider. In this case, both lines receive inbound calls to this VoIP account. The PSTN Line should be configured with a sufficiently long answer delay before the call is automatically answered to allow for the function of the PSTN gateway.

If the service provider does not allow more than one register contact, the PSTN Line should not register. In this case, only Line 1 rings on the inbound call to this VoIP account because it is the only line registered with the service provider.

Line 1 can have the call forwarded to the PSTN Line after a few seconds using the Call-Forward-On-No-Answer feature with gw0 as the forward destination. Similarly, Line 1 can apply Call-Forward-All, Call-Forward-On-Busy, and Call-Forward-Selective feature, and direct the caller to the PSTN-Gateway.

Only PIN authentication is allowed when a VoIP caller is forwarded to the PSTN-gateway from Line 1. If HTTP Authentication is used, the caller is not authenticated.

Another option when using the Forward-To-GW0 feature is to forward the caller to a specific PSTN number, using the syntax <PSTN-number>@gw0 in the forward destination. When using this with Call-Forward-Selective, you can develop some interesting applications. For example, you can forward all callers with 408 area code to 14081234567, or all callers with 800 area code to 18005558355 (This is the number for Tell Me). When this syntax is used, authentication is not used and the target PSTN number is automatically dialed by the SPA3102 after the caller is forwarded to gw0.

Other Options

This section describes other options provided by the SPA3102. It includes the following topics:

• PSTN Call to Ring Line 1, page 4-8

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• Symmetric RTP, page 4-8

• Call Progress Tones, page 4-8

PSTN Call to Ring Line 1

This feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops. After a given number of rings, the VoIP gateway picks up the call.

Symmetric RTP

Symmetric RTP is used to send audio RTP to the source IP and port of the inbound RTP packets. This facilitates NAT traversal.

Table 1-33 lists symmetric RTP parameters.

Table 1-33 Symmetric RTP Parameters

Chapter 4 Configuring the PSTN Gateway (FXO)

Parameter Group Description Range

Symmetric RTP Line 1 Enable symmetric RTP operation. If enabled, SPA3102

Symmetric RTP PSTN

Call Progress Tones

Table 1-34 Call Progress Tones

Call Progress Tone Description

VoIP PIN Tone This tone is played to prompt a VoIP caller to enter a PIN number.

PSTN PIN Tone This tone is played to prompt a PSTN caller to enter a PIN number.

Outside Dial Tone During two-stage PSTN-gateway dialing and with a dial plan assigned, the

Yes or sends RTP packets to the source address of the last received valid inbound RTP packet. If disabled, SPA3102 sends RTP to the destination as indicated in the inbound SDP. Default is yes.

Same as above. Default is yes. Yes or

Line

SPA3102 collects digits from the VoIP caller and processes the number using the dial plan. The SPA3102 plays the <Outside Dial Tone> to prompt the VoIP caller to enter the PSTN number. This tone should be specified to sound different from the PSTN dial tone.

Call Scenarios

This section describes some typical scenarios where the SPA3102 can be applied. Some terms are introduced in the first few sections and reused in later sections. This section includes the following topics:

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• PSTN to VoIP Call with and Without Ring-Thru, page 4-9

• VoIP to PSTN Call with and Without Authentication, page 4-9

• Call Forwarding to PSTN Gateway, page 4-10

• User Dialing 9 to Access PSTN-Gateway for Local Calls, page 4-11

• Using the PSTN-Gateway for 311 and 911 Calls, page 4-12

• Auto-Fallback to the PSTN-Gateway, page 4-12

PSTN to VoIP Call with and Without Ring-Thru

The PSTN caller calls the PSTN line connected to the FXO port. Ring-Thru is disabled. After it rings for a delay equals to the value in <PSTN Answer Delay>, the VoIP gateway answers the call and prompts the PSTN caller to enter a PIN number (assuming PIN authentication is enabled). After a valid PIN is entered, a regular dial plan is played to prompt the PSTN caller to dial the VoIP number. A dial plan is selected according to the PIN number entered by the caller. If authentication is disabled, the default PSTN dial plan is used. Note than the dial plan choice cannot be 0 for a PSTN caller.

Call Scenarios

Note A <PSTN Access List> in terms of Caller ID (ANI) patterns can be configured into the SPA3102 to

automatically grant access to the PSTN caller without entering the PIN. In this case, the default PSTN dial plan is also used.

The same scenario can be implemented using Ring-Thru. When the PSTN line rings, Line 1 rings also. This feature is called Ring-Thru. If Line connected to the PSTN call. Line 1 hears a call waiting tone if it is already connected to another call.

1 is picked up before the VoIP gateway auto-answers, it is

VoIP to PSTN Call with and Without Authentication

This section describes three scenarios with and without authentication and includes the following topics:

• Using PIN Authentication, page 4-9

• Using HTTP Digest Authentication, page 4-10

• Without Authentication, page 4-10

Using PIN Authentication

This scenario assumes that the PSTN Line has a different VoIP account than the Line 1 account. The VoIP caller calls the FXO number, which auto-answers after <VoIP Answer Delay>. The SPA3102 then prompts the VoIP caller for a PIN. When a valid PIN is entered, the SPA3102 plays the <Outside Dial Tone> and prompts the caller to dial the PSTN number.

The number dialed is processed by the dial plan corresponding to the VoIP caller. If the dial plan choice is 0, no dial plan is needed and the user hears the PSTN dial tone right after the PIN is entered. If the dial plan choice is not 0, the final number returned from the dial plan after the complete number is dialed by the caller is dialed to the PSTN. The caller does not hear the PSTN dial tone (except for a little leakage before the first digit of the final number is auto-dialed by the SPA3102).

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If the PSTN Line is busy (off-hook, ringing, or PSTN line not connected) when the VoIP caller calls, the SPA3102 replies with 503. If the PIN number is invalid or entered after the VoIP call leg is connected, the SPA3102 plays the reorder tone to the VoIP caller and eventually ends the call when the reorder tone times out.

Note If <VoIP Caller ID Pattern> is specified and the VoIP caller ID does not match any of the given patterns,

the SPA3102 rejects the call with a 403. This rule applies regardless of the authentication method, even when the source IP address of the INVITE request is in the <VoIP Access List>.

Using HTTP Digest Authentication

The same scenario can be implemented with HTTP digest authentication when the calling device supports the configuration of a auth-ID and password to access the SPA3102 PSTN gateway. When the VoIP caller calls the PSTN Line, the SPA3102 challenges the INVITE request with a 401 response. The calling device should then provide the correct credentials in a subsequent retry of the INVITE, computed with the auth-ID and password using MD5.

If the credentials are correct, the target number specified in the user-id field of the INVITE Request-URI is processed by the dial plan corresponding to the VoIP user (assuming the dial plan choice is not 0). The final number is then auto-dialed by the SPA3102.

Chapter 4 Configuring the PSTN Gateway (FXO)

If the credentials are incorrect, the SPA3102 challenges the INVITE again. If the auth-ID does not exist in the SPA3102 configuration, the SPA3102 replies 403 to the INVITE. If the target number is invalid according to the corresponding dial plan, the SPA3102 also replies 403 to the INVITE. Again, if the PSTN Line is busy at the time of the call, the SPA3102 replies 503.

Note HTTP Digest Authentication is one way to perform one-stage dialing of a VoIP-To-PSTN call. The other

way is with no authentication require. However, if the target number is not specified in the Request-URI or the number matches the account user-id of the PSTN Line, the call reverts to two-stage dialing.

Without Authentication

This scenario can also be implemented without authentication, using one-stage or two-stage dialing, as in the HTTP Authentication case. The default VoIP caller dial plan is used in this scenario. Authentication is performed when the method is none or when the source IP address of the inbound INVITE matches one of the <VoIP Access List> patterns.

Call Forwarding to PSTN Gateway

This section describes a number of scenarios that forward calls to the PSTN gateway. It includes the following topics:

• Forward-On-No-Answer to the PSTN Gateway, page 4-11

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• Forward-All to the PSTN gateway, page 4-11

• Forward to a Particular PSTN Number, page 4-11

• Forward-On-Busy to PSTN Gateway or Number, page 4-11

• Forward-Selective to PSTN Gateway or Number, page 4-11

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Forward-On-No-Answer to the PSTN Gateway

In this scenario, Line 1 is configured to <Forward-On-No-Answer> to the PSTN Gateway. The scenario is implemented by setting User 1 to forward to gw0 on no answer, with <No Answer Delay> set to six seconds.

The caller calls Line 1 and if Line 1 is not picked up after six seconds, the PSTN Line picks up the call and the call reverts to a PSTN-Gateway call, as described above. In this case, HTTP authentication is not allowed because Line 1 does not authenticate inbound INVITE requests. If you need to authenticate the VoIP caller in this case, you must select the PIN authentication method, or else the caller is not authenticated.

Note If the PSTN Line is busy at the moment of the forward, it does not answer the VoIP call. The call forward

rule is ignored and Line 1 continues to ring.

Forward-All to the PSTN gateway

In this scenario, Line 1 is configured with <Forward-All> to the PSTN gateway.This scenario is the same the previous case, except the FXO picks up the Line 1 call immediately.

If the PSTN Line is busy at the moment of the call, the PSTN Line does not pick up the call, the call forward rule is ignored, and Line 1 continues to ring.

Call Scenarios

Forward to a Particular PSTN Number

In this scenario, the forward destination is set to <target-number>@gw0>. This is the same as in the previous examples , except that the SPA3102 automatically dials the given target number on the PSTN line right after it answers the VoIP call leg. This is a special case of one-stage dialing where the target number is specified in the configuration. The caller is not authenticated in this case regardless of the authentication method. However, the caller is still limited by the <VoIP Caller ID Pattern> parameter

Forward-On-Busy to PSTN Gateway or Number

This scenario is similar to the previous cases of call forwarding to gw0, but this applies when Line 1 is active.

Forward-Selective to PSTN Gateway or Number

This scenario is similar to the previous cases of call forwarding to gw0, but this applies when the caller matches the specific caller-id pattern.

User Dialing 9 to Access PSTN-Gateway for Local Calls

To implement this scenario, add the rule “<9,:1408>xxxxxxx<:@gw0>” to the Line 1 dial plan. When user dials 9, SPA3102 plays outside dial tone. The user then dials seven digits and the SPA3102 prepends 1408 before dialing the final number on the PSTN line.

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Using the PSTN-Gateway for 311 and 911 Calls

To implement this scenario, add the rule “[39]11<:@gw0>” to Line 1. When the user dials 311 or 911, the call is routed to the PSTN gateway.

Note If the PSTN Line is busy after the user dials 311 or 911, the call still fails. For true life-line supports,

therefore, the PSTN Line cannot be shared.

Auto-Fallback to the PSTN-Gateway

To implement this scenario, enable <Auto PSTN Fallback>. When registration fails or link is down, the SPA3102 automatically calls “fallback@gw0” when user picks up Line 1. The SPA3102 does not reboot when the link is down. However, the SPA3102 reboots when the link is back up and Line 1 and PSTN Line are not in use.

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CHA PTER

Linksys ATA Field Reference

This chapter describes the fields within each section of the following administration web server pages:

• Info Tab, page 5-2

• System Tab, page 5-8

• SIP Tab, page 5-11

• Regional Tab, page 5-19

• Line Tab, page 5-33

• PSTN Line Tab, page 5-49

• User 1/2 Tab, page 5-65

• PSTN User Tab (SPA3102/AG310), page 5-70

Note Througout this chapter, references to the SPA3102 also apply to the AG310. The AG310 provides the

same functionality as the SPA3102, as well as an ADSL modem and a four-port Ethernet switch.

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Info Tab

This section describes the fields for the following headings on the Info tab:

• System Information (PAP2T), page 5-2

• System Status (VoIP), page 5-2

• Line 1/2 Status, page 5-3

• PSTN Line Status, page 5-5

Note The fields on this tab are read-only and cannot be edited.

System Information (PAP2T)

Field Description

DHCP Indicates if DHCP is enabled.

Chapter 5 Linksys ATA Field Reference

Current IP Displays the current IP address assigned to the Linksys IP phone.

Host Name Displays the current IP address assigned to the Linksys IP phone.

Domain Displays the network domain name of the Linksys IP phone.

Current Netmask Displays the network mask assigned to the Linksys IP phone.

Current Gateway Displays the default router assigned to the Linksys IP phone.

Primary DNS Displays the primary DNS server assigned to the Linksys IP phone.

Secondary DNS Displays the secondary DNS server assigned to the Linksys IP phone.

System Status (VoIP)

Field Description

Current Time Current date and time of the system; for example, 10/3/2003 16:43:00.

Elapsed Time Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.

Broadcast Pkts Sent Total number of broadcast packets sent.

Broadcast Bytes Sent Total number of broadcast packets received.

Broadcast Pkts Recv Total number of broadcast bytes sent.

Broadcast Bytes Recv Total number of broadcast bytes received and processed.

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Field Description

Current Time Current date and time of the system; for example, 10/3/2003 16:43:00.

Broadcast Pkts Dropped Total number of broadcast packets received but not processed.

Broadcast Bytes Dropped Total number of broadcast bytes received but not processed.

RTP Packets Sent Total number of RTP packets sent (including redundant packets).

RTP Bytes Sent Total number of RTP packets received (including redundant packets).

RTP Packets Recv Total number of RTP bytes sent.

RTP Bytes Recv Total number of RTP bytes received.

SIP Messages Sent Total number of SIP messages sent (including retransmissions).

SIP Bytes Sent Total number of SIP messages received (including retransmissions).

SIP Messages Recv Total number of bytes of SIP messages sent (including retransmissions).

SIP Bytes Recv Total number of bytes of SIP messages received (including retransmissions).

Info Tab

External IP External IP address used for NAT mapping.

Line 1/2 Status

Field Description

Display Name The value that is shown through Caller ID.

User ID The user ID assigned by the ITSP (often the phone number) used for registering the SIP

user agent with the SIP proxy at the ITSP.

Hazardous Potential The amount of hazardous potential is displayed.

Foreign Voltage The amount of foreign voltage is displayed.

Long Resistive Fault The amount of long resistive fault is displayed.

REN The Ringer Equivalence Number (REN) is displayed.

Hook State Status of phone readiness. On indicates that the phone is ready for use; Off indicates that

the phone is in use.

Registration State Indicates if the line has registered with the SIP proxy.

Last Registration At Last date and time the line was registered.

Next Registration In Number of seconds before the next registration renewal.

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Info Tab

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 Indicates whether a call back request is in progress: Yes or No.

Last Called Number Last number called.

Last Caller Number Number of the last caller.

Mapped SIP Port Port number of the SIP port mapped by NAT.

Call 1/2 State Status of the call.

Call 1/2 Tone Type of tone used by the call.

Call 1/2 Encoder Codec used for encoding.

Call 1/2 Decoder Codec used for decoding.

Call 1/2 FAX Status of the fax pass-through mode.

Call 1/2 Type Direction of the call.

Call 1/2 Remote Hold Indicates whether the far end has placed the call on hold.

Call 1/2 Callback Indicates whether the call was triggered by a call back request.

Call 1/2 Peer Name Name of the internal phone.

Call 1/2 Peer Phone Phone number of the internal phone.

Call 1/2 Duration Duration of the call.

Call 1/2 Packets Sent Number of packets sent.

Call 1/2 Packets Recv Number of packets received.

Call 1/2 Bytes Sent Number of bytes sent.

Call 1/2 Bytes Recv Number of bytes received.

Call 1/2 Decode Latency Number of milliseconds for decoder latency.

Call 1/2 Jitter Number of milliseconds for receiver jitter.

Call 1/2 Round Trip Delay Number of milliseconds for delay.

Call 1/2 Packets Lost Number of packets lost.

Call 1/2 Packet Error Number of invalid packets received.

Call 1/2 Mapped RTP Port The port mapped for Real Time Protocol traffic for Call 1/2.

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PSTN Line Status

Note References to the SPA3102 also apply to the AG310.

Field Description

(PSTN) Hook State Hook state of the FXO port. Either On or Off.

(PSTN) Line Voltage The voltage existing on the PSTN line.

(PSTN) Loop Current The current (milliamperes) existing on the local loop.

Registration State Indicates if the line has registered with the SIP proxy.

Last Registration At Last date and time the line was registered.

Next Registration In Number of seconds before the next registration renewal.

Last Called VoIP Number The last VoIP number called from the FXO Line.

Info Tab

Last Called PSTN Number The PSTN number dialed by the SPA (logged only if a non-trivial dial plan is used).

Last VoIP Caller The last VoIP caller to the FXO Line.

Last PSTN Caller Name and number of the last PSTN caller.

Last PSTN Disconnect Reason Reason for SPA hanging up the FXO port. Can be one of the following:

• PSTN Disconnect Tone

• PSTN Activity Timeout

• CPC Signal

• Polarity Reversal

• Vo I P C a l l F a i l e d

• VoIP Call Ended

• Invalid VoIP Destination

• Invalid PIN

• PIN Digit Timeout

• VoIP Dialing Timeout

• PSTN Gateway Call Timeout

• VoIP Gateway Call Timeout

PSTN Activity Timer Shows the time (ms) before the SPA disconnects the current gateway unless the PSTN side

has some audio activity.

Mapped SIP Port Same as Line 1.

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Info Tab

Call Type May take one of the following values:

• PSTN Gateway Call = VoIP-To-PSTN Call

• VoIP Gateway Call = PSTN-To-VoIP Call

• PSTN To Line 1 = PSTN call ring through and answered by Line 1

• Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW

• Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number

• Line 1 To PSTN Gateway

• Line 1 Fallback To PSTN Gateway

Vo I P S t a t e Same as Line 1 Call 1.

PSTN State May take one of the following values:

• Idle

• Collecting PSTN Pin

• Invalid PSTN PIN

• PSTN Caller Accepted

• Connected to PSTN

Chapter 5 Linksys ATA Field Reference

Vo I P To n e Same as Line 1 Call 1 (Indicate what tone is being played to the VoIP call leg).

PSTN Tone Indicate what tone is being played to the PSTN call leg.

VoIP Peer Name Same as Line 1 Call 1 (Name of the party at the VoIP call leg).

PSTN Peer Name Name of the party at the PSTN call leg.

VoIP Peer Number Same as Line 1 Call 1 (Phone number of the party at the VoIP call leg).

PSTN Peer Number Phone number of the party at the PSTN call leg.

VoIP Call Encoder Same as Line 1 Call 1 (Audio encoder being used for the VoIP call leg).

VoIP Call Decoder Same as Line 1 Call 1 (Audio decoder being used for the VoIP call leg).

VoIP Call FAX Same as Line 1 Call 1.

VoIP Call Remote Hold Same as Line 1 Call 1.

VoIP Call Duration Same as Line 1 Call 1.

VoIP Call Packets Sent Same as Line 1 Call 1.

VoIP Call Packets Recv Same as Line 1 Call 1.

VoIP Call Bytes Sent Same as Line 1 Call 1.

VoIP Call Bytes Recv Same as Line 1 Call 1.

VoIP Call Decode Latency Same as Line 1 Call 1.

VoIP Call Jitter Same as Line 1 Call 1.

VoIP Call Round Trip Delay Same as Line 1 Call 1.

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VoIP Call Packets Lost Same as Line 1 Call 1.

VoIP Call Packet Error Same as Line 1 Call 1.

VoIP Call Mapped RTP Port Same as Line 1 Call 1.

Info Tab

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System Tab

This section describes the fields for the following headings on the System tab:

• System Configuration, page 5-8

• Internet Connection Type (PAP2T), page 5-8

• Optional Network Configuration (PAP2T), page 5-9

• Miscellaneous Settings (Not in PAP2T), page 5-10

System Configuration

Field Description

Restricted Access Domains This feature is used when implementing software customization.

Enable Web Server Enable/disable web server of Linksys IP phone

This feature should only be used on firmware version 1.0.9 or later.

The default is yes.

Web Server Port Port number of the Linksys IP phone administration web server.

The default is 80.

Enable Web Admin Access 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 Password for the administrator.

The default is no password.

User Password Password for the user.

The default is no password.

Internet Connection Type (PAP2T)

Field Description

DHCP Enable or disable DHCP.

The default is yes.

Static IP Static IP address of Linksys IP phone, which takes effect if DHCP is disabled.

The default is 0.0.0.0.

NetMask The NetMask used by Linksys IP phone when DHCP is disabled.

The default is 255.255.255.0.

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Field Description

Gateway The default gateway used by Linksys IP phone when DHCP is disabled.

The default is 0.0.0.0.

Optional Network Configuration (PAP2T)

Field Description

Host Name The host name of the Linksys IP phone.

Domain The network domain of the Linksys IP phone.

Primary DNS DNS server used by Linksys IP phone in addition to DHCP supplied DNS servers if

DHCP is enabled; when DHCP is disabled, this is the primary DNS server.

The default is 0.0.0.0.

Secondary DNS DNS server used by Linksys IP phone in addition to DHCP supplied DNS servers if

DHCP is enabled; when DHCP is disabled, this is the secondary DNS server.

The default is 0.0.0.0.

DNS Server Order Specifies the method for selecting the DNS server. The options are Manual,

Manual/DHCP, and DHCP/Manual.

DNS Query Mode Do parallel or sequential DNS Query. With parallel DNS query mode, the Linksys IP

phone sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the Linksys IP phone.

The default is parallel.

Syslog Server Specify the syslog server name and port. This feature specifies the server for logging

Linksys IP phone system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.

Debug Server The debug server name and port. This feature specifies the server for logging Linksys IP

phone debug information. The level of detailed output depends on the debug level parameter setting.

Debug Level The higher the debug level, the more debug information is generated. Zero (0) means no

debug information is generated. To log SIP messages, Debug Level must be set to at least

The default is 0.

Primary NTP Server IP address or name of primary NTP server.

Secondary NTP Server IP address or name of secondary NTP server.

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Miscellaneous Settings (Not in PAP2T)

Field Description

Syslog Server Specifies the IP address of the syslog server.

Debug 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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SIP Tab

This section describes the fields for the following headings on the SIP tab:

• SIP Parameters, page 5-11

• SIP Timer Values (sec), page 5-12

• Response Status Code Handling, page 5-14

• RTP Parameters, page 5-14

• SDP Payload Types, page 5-15

• NAT Support Parameters, page 5-16

SIP Parameters

Field Description

Max Forward SIP Max Forward value, which can range from 1 to 255.

The default is 70.

SIP Tab

Max Redirection Number of times an invite can be redirected to avoid an infinite loop.

The default is 5.

Max Auth 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 Linksys IP phone 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.

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.

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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 Linksys IP phone uses compact SIP headers in outbound SIP messages. If set to no, the Linksys IP phone uses normal SIP headers. If inbound SIP requests contain compact headers, Linksys IP phone 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, Linksys IP phone 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 Lets you keep the Display Name private. Select yes if you want the Linksys IP phone 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.

RFC 2543 Call Hold

The default is no.

Mark All AVT Packets

SIP TCP Port Min

SIP TCP Port Min Specifies the lowest TCP port number that can be used for SIP sessions.

SIP TCP Port Max Specifies the highest TCP port number that can be used for SIP sessions.

CTI Enable 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 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.

SIP T4 RFC 3261 T4 value (maximum duration a message remains in the network), which can

range from 0 to 64 seconds.

The default is 5.

SIP Timer B INVITE time-out value, which can range from 0 to 64 seconds.

The default is 32.

SIP Timer F Non-INVITE time-out value, which can range from 0 to 64 seconds.

The default is 32.

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SIP Timer H INVITE final response, time-out value, which can range from 0 to 64 seconds.

The default is 32.

SIP Timer D ACK hang-around time, which can range from 0 to 64 seconds.

The default is 32.

SIP Timer J 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 Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request.

The default is 30. Range: 0–(2

Reg Min Expires 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 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.

Reg Retry Intvl Interval to wait before the Linksys IP phone 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 Linksys IP phone waits for the specified length of time before retrying. If this interval is 0, the Linksys IP phone 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.

–1).

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.

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.

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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.

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.

SIT3 RSC SIP response status code to INVITE on which to play the SIT3 Tone.

SIT4 RSC SIP response status code to INVITE on which to play the SIT4 Tone.

Try Backup RSC SIP response code that retries a backup server for the current request.

Retry Reg RSC Interval to wait before the Linksys IP phone retries registration after failing during the last

registration.

The default is 30.

Reorder or Busy Tone is played by default for all

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.

Max RTP ICMP Err Number of successive ICMP errors allowed when transmitting RTP packets to the peer

before the Linksys IP phone terminates the call. If value is set to 0, the Linksys IP phone ignores the limit on ICMP errors.

The default is 0.

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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 Linksys IP phone 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/Linksys IP phone-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the Linksys IP phone’s local time, not the time reported by an NTP server. If the Linksys IP phone 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 Linksys IP phone web page.

The default is 0.

No UDP Checksum Select yes if you want the Linksys IP phone to calculate the UDP header checksum for

SIP messages. Otherwise, select no.

The default is no.

Stats In BYE Determines whether the Linksys IP phone 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

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.

INFOREQ Dynamic Payload INFOREQ dynamic payload type.

There is no default.

G726r16 Dynamic Payload G.726-16 dynamic payload type. The valid range is 96-127.

The default is 98.

G726r24 Dynamic Payload G.726-24 dynamic payload type. The valid range is 96-127.

The default is 97.

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G726r40 Dynamic Payload G.726-40 dynamic payload type. The valid range is 96-127.

The default is 96.

G729b Dynamic Payload G.729b dynamic payload type. The valid range is 96-127.

The default is 99.

NSE Codec Name 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 G.726-16 codec name used in SDP.

The default is G726-16.

G726r24 Codec Name G.726-24 codec name used in SDP.

The default is G726-24.

G726r32 Codec Name G.726-32 codec name used in SDP.

The default is G726-32.

G726r40 Codec Name G.726-40 codec name used in SDP.

The default is G726-40.

G729a Codec Name G.729a codec name used in SDP.

The default is G729a.

G729b Codec Name G.729b codec name used in SDP.

The default is G729ab.

G723 Codec Name G.723 codec name used in SDP.

The default is G723.

EncapRTP Codec Name EncapRTP codec name used in SDP.

The default is EncapRTP.

NAT Support Parameters

Field Description

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Frequently Asked Questions

User Manual