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Linksys SPA1001 Administrator User Manual
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Linksys SPA1001 Administrator User Manual

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Linksys ATA Administrator Users Guide
Document Version 3.2
Corporate Headquarters
Linksys
121 Theory Drive
Irvine, CA 92617
USA
http://www.linksys.com
Tel: 949 823-1200
800 546-5797
Fax: 949 823-1100

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  • Page 1 Linksys ATA Administrator Users Guide Document Version 3.2 Corporate Headquarters Linksys 121 Theory Drive Irvine, CA 92617 http://www.linksys.com Tel: 949 823-1200 800 546-5797 Fax: 949 823-1100...
  • Page 2 Linksys ATA Administrator Guide Copyright ©2007 Cisco Systems, Inc. All rights reserved.Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names are trademarks or registered trademarks of their respective holders.

  • Page 3: Table Of Contents

    C O N T E N T S Preface Document Audience i-xi Linksys Analog Telephone Adapters i-xi How This Document is Organized i-xii Document Conventions i-xii Related Documentation i-xiii Technical Support i-xiii i-xiii Introducing Linksys Analog Telephone Adapters C H A P T E R Overview Ensuring Voice Quality Audio Compression Algorithm...
  • Page 4 1-12 Where to Go From Here 1-13 Getting Started C H A P T E R Linksys Analog Telephone Adapters (ATAs) Caring for Your Hardware AG310 PAP2T RTP300 SPA1001 SPA2102 SPA3102 SPA8000 WRP400 2-10 WRTP54G 2-11 WRT54GP2 2-13 Establishing Connectivity...
  • Page 5 Contents Provisioning Capabilities Configuration Profile Configuring a Dial Plan Dial Plan Digit Sequences Dial Plan Rules Digit Sequence Syntax Element Repetition Sub-sequence Substitution Inter-sequence Tones Number Barring Interdigit Timer Master Override Local Timer Overrides Pause Dial Plan Examples Dial Plan Timers Interdigit Long Timer Interdigit Short Timer Dial Plans...
  • Page 6 Contents Two-Stage Dialing How PSTN-To-VoIP Calls Work Terminating Gateway Calls VoIP Outbound Call Routing Configuring VoIP Failover to PSTN Sharing One VoIP Account Between the FXS and PSTN Lines Other Options PSTN Call to Ring Line 1 Symmetric RTP Call Progress Tones Call Scenarios PSTN to VoIP Call with and Without Ring-Thru VoIP to PSTN Call with and Without Authentication...
  • Page 7 Contents SIP Timer Values (sec) 5-12 Response Status Code Handling 5-14 RTP Parameters 5-14 SDP Payload Types 5-15 NAT Support Parameters 5-16 Regional Tab 5-19 Call Progress Tones 5-19 Distinctive Ring Patterns 5-21 Distinctive Call Waiting Tone Patterns 5-21 Distinctive Ring/CWT Pattern Names 5-22 Ring and Call Waiting Tone Spec 5-23...
  • Page 8 Contents Dial Plans 5-57 VoIP-To-PSTN Gateway Setup 5-57 VoIP Users and Passwords (HTTP Authentication) 5-59 Ring Settings 5-60 FXO (PSTN) Timer Values (sec) 5-60 PSTN Disconnect Detection 5-62 International Control (Settings) 5-63 User 1/2 Tab 5-65 Call Forward Settings 5-65 Selective Call Forward Settings 5-66 Speed Dial Settings...
  • Page 9 Contents Call Hold Three-Way Calling Three-Way Ad-Hoc Conference Calling Call Return Automatic Call Back Call FWD—Unconditional Call FWD – Busy C-10 Call FWD—No Answer C-11 Anonymous Call Blocking C-11 Distinctive/Priority Ringing and Call Waiting Tone C-12 Speed Calling—Up to Eight Numbers or IP Addresses C-12 N D E X Linksys ATA Administrator Guide...
  • Page 10 Contents Linksys ATA Administrator Guide Document Version 3.1...
  • Page 11 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 • SPA2102—Voice adapter with router • Linksys ATA Administrator Guide...
  • Page 12 Preface 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 •...
  • Page 13 Preface Related Documentation Related Documentation 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 other Linksys Voice System products: SPA9000 Administrator Guide •...
  • Page 14 Preface Linksys ATA Administrator Guide Document Version 3.1...

  • Page 15: Overview

    (WAN) Ethernet (LAN) Voice Lines Description 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) —...

  • Page 16: C H A P T E R 1 Introducing Linksys Analog Telephone Adapters

    Chapter 1 Introducing Linksys Analog Telephone Adapters Overview 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 •...

  • Page 17: Ensuring Voice Quality

    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.

  • Page 18: Audio Compression Algorithm

    Chapter 1 Introducing Linksys Analog Telephone Adapters 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.

  • Page 19: Hardware Noise

    Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions 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 Hz noise from the power adaptor. The Linksys ATA hardware design minimizes noise coupling.

  • Page 20: Supported Codecs

    Chapter 1 Introducing Linksys Analog Telephone Adapters 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.

  • Page 21: Silence Suppression And Comfort Noise Generation

    Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions 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.”...

  • Page 22: Other Features

    Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions 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.

  • Page 23: Technology Background

    Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background Table 1-4 Linksys ATA Features Feature Description Signaling Hook Flash The Linksys ATA can signal hook flash events to the remote party on a Event connected call. This feature can be used to provide advanced mid-call services with third-party-call-control.

  • Page 24: Session Initiation Protocol

    Chapter 1 Introducing Linksys Analog Telephone Adapters 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. Figure 1-2 illustrates a SIP request for connection to another subscriber in the network.

  • Page 25: Nat Types

    Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background 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.

  • Page 26: Simple Traversal Of Udp Through Nat

    Chapter 1 Introducing Linksys Analog Telephone Adapters 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.

  • Page 27: Where To Go From Here

    Chapter 1 Introducing Linksys Analog Telephone Adapters Where to Go From Here 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.
  • Page 28 Chapter 1 Introducing Linksys Analog Telephone Adapters Where to Go From Here SPA900 Series IP Phones Administrator Guide • Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment • Linksys ATA Administrator Guide 1-14 Document Version 3.1...
  • Page 29 C H A P T E R 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 •...

  • Page 30: Linksys Analog Telephone Adapters (Atas)

    Chapter 2 Getting Started 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.

  • Page 31: Pap2T

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Table 1-5 AG310 Front Panel 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.

  • Page 32: Rtp300

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-5 PAP2T 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 Function Phone 1/2 Steady green indicates active or registered connection to the ITSP through the Phone port.

  • Page 33: Spa1001

    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 The SPA1001 provides one FXS port (see Figure 1-7). Linksys ATA Administrator Guide Document Version 3.1...

  • Page 34: Spa2102

    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.

  • Page 35: Spa3102

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) 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 Function Power Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade.
  • Page 36 Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-9 SPA3102 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 Function Power Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade.

  • Page 37: Spa8000

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) SPA8000 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.

  • Page 38: Wrp400

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Table 1-17 SPA8000 Back Panel Port Function Ethernet maintenance port. Connect to a network device, such as a PC or a switch with an Ethernet cable for accessing the administration web server on the SPA8000.

  • Page 39: Wrtp54G

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) The following tables describe the LEDS on the front panel and the ports on the back panel of the device WRP400 Table 1-18 Front Panel Function Ethernet 1-4 Steady green indicates an active connection to the network. Flashing indicates traffic.
  • Page 40 Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-12 WRTP54G 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 Function Ethernet 1-4 Steady green indicates an active connection to the network.

  • Page 41: Wrt54Gp2

    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) WRT54GP2 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.

  • Page 42: Establishing Connectivity

    Chapter 2 Getting Started 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 •...

  • Page 43: Making The Physical Connections

    Chapter 2 Getting Started Establishing Connectivity 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 •...

  • Page 44: Connecting The Spa8000

    Chapter 2 Getting Started Connecting the SPA8000 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 •...

  • Page 45: Connectivity Requirements

    Chapter 2 Getting Started 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.

  • Page 46: Using The Interactive Voice Response Interface

    Chapter 2 Getting Started Using the Interactive Voice Response Interface 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.

  • Page 47: Ivr Options

    Chapter 2 Getting Started 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.
  • Page 48 Chapter 2 Getting Started Using the Interactive Voice Response Interface Table 1-25 IVR Options (continued) Set Network Mask Enter value using numbers DHCP must be “Disabled,” otherwise on the telephone key pad. you hear, “Invalid Option,” if you try to Use the * (star) key when set this value.

  • Page 49: Entering A Password Through The Ivr

    Chapter 2 Getting Started Using the Interactive Voice Response Interface Table 1-25 IVR Options (continued) User Factory Reset of 877778 Enter 1 to confirm Linksys ATA prompts for confirmation. Unit Enter *(star) to cancel After confirming, you hear “Option operation Successful.”...

  • Page 50: Using The Administration Web Server

    Chapter 2 Getting Started Using the Administration Web Server For example, to input password test#@1234 by phone keypad, you need to press the following sequence of digits: 8378001234. After entering a value, press the # (pound) key to indicate end of input. To save value, press 1.

  • Page 51: Administrator Account Privileges

    Chapter 2 Getting Started Using the Administration Web Server 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.
  • Page 52 Chapter 2 Getting Started Using the Administration Web Server Linksys ATA Administrator Guide 2-24 Document Version 3.1...

  • Page 53: Initial Configuration

    C H A P T E R 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 •...

  • Page 54: Chapter 3 Configuring Linksy Ata

    Chapter 3 Configuring Linksys ATAs Initial Configuration If you use a cable modem, you may need to configure the MAC Clone Settings. (Contact your ISP for more information.) If your service uses a specific PC MAC address, then select yes from the Enable MAC Clone Service setting.

  • Page 55: Web Interface Urls

    Chapter 3 Configuring Linksys ATAs Web Interface URLs 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.

  • Page 56: Reboot Url

    Chapter 3 Configuring Linksys ATAs 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 •...

  • Page 57: Configuring A Dial Plan

    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan <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.

  • Page 58: Dial Plan Rules

    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan 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 •...

  • Page 59: Element Repetition

    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan – 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”, …...

  • Page 60: Dial Plan Examples

    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan 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.

  • Page 61: Dial Plan Timers

    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan 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 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).

  • Page 62: Secure Call Implementation

    Chapter 3 Configuring Linksys ATAs Secure Call Implementation 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 •...

  • Page 63: Using A Mini-Certificate

    Chapter 3 Configuring Linksys ATAs Secure Call Implementation • 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.

  • Page 64: Configuring A Streaming Audio Server

    Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server 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 •...

  • Page 65: Using A Streaming Audio Server

    Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server 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 If the Phone 2 on the PAP2T) and use it as a streaming audio source device. Linksys ATA has multiple can be configured FXS ports, either or both of the associated lines (Line 1 and Line 2 on the PAP2T)

  • Page 66: Example Sas With Moh

    Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server 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.

  • Page 67: Configuring The Streaming Audio Server

    Chapter 3 Configuring Linksys ATAs Using a FAX Machine with the SPA2102 or SPA8000 SAS Enable[1] = no MOH Server [1] = 1002@192.168.2.100:5061 or 1002@127.0.0.1:5061 SAS Enable[2] = yes Linksys ATA 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...
  • Page 68 Preferred Codec: G.711 • Use pref. codec only: yes • If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax Step 4 using modem passthrough. For example: modem passthrough nse payload-type 110 codec g711ulaw...

  • Page 69: Managing Caller Id Service

    Chapter 3 Configuring Linksys ATAs Managing Caller ID Service • 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.

  • Page 70: Troubleshooting And Configuration Faq

    Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ 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 DTMF Ring c) ETSI Onhook Pre-Ring FSK/DTMF Polarity DTMF/ First Reversal (DTAS) Ring...
  • Page 71 Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ 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.
  • Page 72 Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ 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>...

  • Page 73: Overview

    C H A P T E R 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: •...

  • Page 74: Chapter 4 Configuring The Pstn Gateway (Fxo)

    Chapter 4 Configuring the PSTN Gateway (FXO) How VoIP-To-PSTN Calls Work 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.

  • Page 75: Two-Stage Dialing

    Chapter 4 Configuring the PSTN Gateway (FXO) How VoIP-To-PSTN Calls Work Table 1-27 VoIP User Account Information Parameter Group Description Values User ID PSTN The username value. 31-character string 1/2/3/4/5/6/7/ Line Password PSTN The password value. 31-character string 1/2/3/4/5/6/7/ Line User PSTN Specifies the dial plan to be used for this VoIP...

  • Page 76: How Pstn-To-Voip Calls Work

    Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work Table 1-28 Two-Stage Dialing Parameter Group Description Values VoIP Caller PSTN The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8. 31-character string 1/2/3/4/5/6/7/8 Line VoIP Caller PSTN Specifies which dial plan to be used for this...

  • Page 77: Voip Outbound Call Routing

    Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work Table 1-29 Terminating Gateway Call Parameters Parameter Group Description Values Detect CPC: PSTN Yes or No If yes, SPA3102 detects CPC as a disconnect signal. Line Default = Yes Detect Long Silence: PSTN If yes, SPA3102 detects prolonged silence period as...
  • Page 78 Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work 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 Domain name or IP gateway. If the port number is not specified, 5060 is address assumed.

  • Page 79: Configuring Voip Failover To Pstn

    Chapter 4 Configuring the PSTN Gateway (FXO) Configuring VoIP Failover to PSTN 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.

  • Page 80: Symmetric Rtp

    Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios 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.

  • Page 81: Pstn To Voip Call With And Without Ring-Thru

    Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios • 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 •...

  • Page 82: Using Http Digest Authentication

    Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios 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.

  • Page 83: Forward-On-No-Answer To The Pstn Gateway

    Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios 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.

  • Page 84: Using The Pstn-Gateway For 311 And 911 Calls

    Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios 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. If the PSTN Line is busy after the user dials 311 or 911, the call still fails.
  • Page 85 C H A P T E R 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 •...

  • Page 86: Chapter 5 Linksy Ata Field Reference

    Chapter 5 Linksys ATA Field Reference Info Tab 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 •...

  • Page 87: Line 1/2 Status

    Chapter 5 Linksys ATA Field Reference Info Tab 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).
  • Page 88 Chapter 5 Linksys ATA Field Reference 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.

  • Page 89: Pstn Line Status

    Chapter 5 Linksys ATA Field Reference Info Tab PSTN Line Status References to the SPA3102 also apply to the AG310. Note 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.
  • Page 90 Chapter 5 Linksys ATA Field Reference 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 •...
  • Page 91 Chapter 5 Linksys ATA Field Reference Info Tab 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. Linksys IP Phone Administrator Guide Document Version 3.2...

  • Page 92: System Tab

    Chapter 5 Linksys ATA Field Reference System Tab 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 •...

  • Page 93: Optional Network Configuration (Pap2T)

    Chapter 5 Linksys ATA Field Reference System Tab 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.

  • Page 94: Miscellaneous Settings (Not In Pap2T)

    Chapter 5 Linksys ATA Field Reference System Tab 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.

  • Page 95: Sip Tab

    Chapter 5 Linksys ATA Field Reference SIP Tab 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 •...

  • Page 96: Sip Timer Values (Sec)

    Chapter 5 Linksys ATA Field Reference SIP Tab 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.
  • Page 97 Chapter 5 Linksys ATA Field Reference SIP Tab 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.

  • Page 98: Response Status Code Handling

    Chapter 5 Linksys ATA Field Reference SIP Tab 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 Reorder or Busy Tone is played by default for all returned, the SIT1 tone is played.

  • Page 99: Sdp Payload Types

    Chapter 5 Linksys ATA Field Reference SIP Tab 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.

  • Page 100: Nat Support Parameters

    Chapter 5 Linksys ATA Field Reference SIP Tab 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.
  • Page 101 Chapter 5 Linksys ATA Field Reference SIP Tab Handle VIA received If you select yes, the Linksys IP phone processes the received parameter in the VIA header (this is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored.
  • Page 102 Chapter 5 Linksys ATA Field Reference SIP Tab EXT RTP Port Min External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range.

  • Page 103: Regional Tab

    Chapter 5 Linksys ATA Field Reference Regional Tab Regional Tab This section describes the fields for the following headings on the Regional tab: Call Progress Tones, page 5-19 • Distinctive Ring Patterns, page 5-21 • Distinctive Call Waiting Tone Patterns, page 5-21 •...
  • Page 104 Chapter 5 Linksys ATA Field Reference Regional Tab Ring Back Tone Played during an outbound call when the far end is ringing. The default is 440@-19,480@-19;*(2/4/1+2). Confirm Tone Brief tone to notify the user that the last input value has been accepted. The default is 600@-16;...

  • Page 105: Distinctive Ring Patterns

    Chapter 5 Linksys ATA Field Reference Regional Tab Distinctive Ring Patterns Field Description Ring1 Cadence Cadence script for distinctive ring 1. The default is 60(2/4). Ring2 Cadence Cadence script for distinctive ring 2. The default is 60(.3/.2, 1/.2,.3/4. Ring3 Cadence Cadence script for distinctive ring 3.

  • Page 106: Distinctive Ring/Cwt Pattern Names

    Chapter 5 Linksys ATA Field Reference Regional Tab CWT6 Cadence Cadence script for distinctive CWT 6. The default is 30(.1/.1, .3/.1, .1/9.3). CWT7 Cadence Cadence script for distinctive CWT 7. The default is 30(.1/.1, .3/.1, .1/9.3). CWT8 Cadence Cadence script for distinctive CWT 8. The default is 2.3(..3/2).

  • Page 107: Ring And Call Waiting Tone Spec

    Chapter 5 Linksys ATA Field Reference Regional Tab Ring and Call Waiting Tone Spec Field Description Ring Waveform Waveform for the ringing signal. The default is Sinusoid. Ring Frequency Frequency of the ringing signal. Valid values are 10–100 (Hz). The default is 25. Ring Voltage Ringing voltage.

  • Page 108: Vertical Service Activation Codes

    Chapter 5 Linksys ATA Field Reference Regional Tab VMWI Refresh Intvl Interval between VMWI refresh to the CPE. The default is 0.5. Interdigit Long Timer Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed.
  • Page 109 Chapter 5 Linksys ATA Field Reference Regional Tab Call Back Act Code Starts a callback when the last outbound call is not busy. The default is *66. Call Back Deact Code Cancels a callback. The default is *86. Call Back Busy Act Code Starts a callback when the last outbound call is busy.
  • Page 110 Chapter 5 Linksys ATA Field Reference Regional Tab CW Deact Code Disables call waiting on all calls. The default is *57. CW Per Call Act Code Enables call waiting for the next call. The default is *71. CW Per Call Deact Code Disables call waiting for the next call.
  • Page 111 Chapter 5 Linksys ATA Field Reference Regional Tab Speed Dial Act Code Assigns a speed dial number. The default is *74. Secure All Call Act Code Makes all outbound calls secure. The default is *16. Secure No Call Act Code Makes all outbound calls not secure.

  • Page 112: Vertical Service Announcement Codes

    Chapter 5 Linksys ATA Field Reference Regional Tab Feature Dial Services Codes These codes tell the Linksys IP phone what to do when the user is listening to the first or second dial tone. One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc.
  • Page 113 Chapter 5 Linksys ATA Field Reference Regional Tab Field Description Prefer G711u Code Makes this codec the preferred codec for the associated call. The default is *017110. Force G711u Code Makes this codec the only codec that can be used for the associated call. The default is *027110.

  • Page 114: Miscellaneous

    Chapter 5 Linksys ATA Field Reference Regional Tab Miscellaneous Field Description Set Local Date (mm/dd) Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits. Set Local Time (HH/mm) Sets the local time (hh stands for hours and mm stands for minutes).
  • Page 115 Chapter 5 Linksys ATA Field Reference Regional Tab FXS Port Input Gain Input gain in dB, up to three decimal places. The range is 6.000 to -12.000. The default is -3. FXS Port Output Gain Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the <FXS Port Output Gain>.
  • Page 116 Chapter 5 Linksys ATA Field Reference Regional Tab Feature Invocation Method Select the method you want to use, Default or Sweden default. (Not in PAP2T) The default is Default. More Echo Suppression Enable or disable more echo suppresion. The default is no. GR909 Test To use this test, select yes.

  • Page 117: Line Tab

    Chapter 5 Linksys ATA Field Reference Line Tab Line Tab This section describes the fields for the following headings on the Line tabs: Line Enable, page 5-33 • Streaming Audio Server (SAS), page 5-33 • NAT Settings, page 5-34 • Network Settings, page 5-35 •...

  • Page 118: Nat Settings

    Chapter 5 Linksys ATA Field Reference Line Tab SAS Enable To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller.

  • Page 119: Network Settings

    Chapter 5 Linksys ATA Field Reference Line Tab Destination that should receive NAT keep alive messages. If the value is $PROXY, NAT Keep Alive Dest the messages are sent to the current or outbound proxy. The default is $PROXY. Network Settings Field Description SIP ToS/DiffServ Value...
  • Page 120 Chapter 5 Linksys ATA Field Reference Line Tab SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. The default is no. EXT SIP Port The external SIP port number.
  • Page 121 Chapter 5 Linksys ATA Field Reference Line Tab SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows: • none—No logging. • 1-line—Logs the start-line only for all messages. •...

  • Page 122: Call Feature Settings

    Chapter 5 Linksys ATA Field Reference Line Tab Refer-To Target Contact To contact the refer-to target, select yes. Otherwise, select no. The default is no. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE.
  • Page 123 Chapter 5 Linksys ATA Field Reference Line Tab Field Description Use Outbound Proxy Enable the use of <Outbound Proxy>. If set to no, <Outbound Proxy> and <Use OB Proxy in Dialog) is ignored. The default is no. Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Use OB Proxy In Dialog Whether to force SIP requests to be sent to the outbound proxy within a dialog.

  • Page 124: Subscriber Information

    Chapter 5 Linksys ATA Field Reference Line Tab Field Description Voice Mail Server Enter the URL or IP address of the server. Mailbox Subscribe Expires <<help here>> Subscriber Information Field Description Display Name Display name for caller ID. User ID Extension number for this line.
  • Page 125 Chapter 5 Linksys ATA Field Reference Line Tab Field Description Call Waiting Serv Enable Call Waiting Service. The default is yes. Block CID Serv Enable Block Caller ID Service. The default is yes. Block ANC Serv Enable Block Anonymous Calls Service The default is yes.

  • Page 126: Audio Configuration

    Chapter 5 Linksys ATA Field Reference Line Tab Field Description Call Back Serv Enable Call Back Service Three Way Calling is required for Three Way Three Way Call Serv Enable Three Way Calling Service. Conference and Attended Transfer. The default is yes. Three Way Conf Serv Enable Three Way Conference Service.
  • Page 127 Chapter 5 Linksys ATA Field Reference Line Tab Therefore it is important to disable the use of G.729a in order to guarantee the support of two simultaneous G.723/G.726 codec. Field Description Preferred Codec Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723.
  • Page 128 Chapter 5 Linksys ATA Field Reference Line Tab G726-40 Enable To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no. The default is yes. FAX Passthru Codec Select the codec for fax passthrough, G711u or G711a. The default is G711u.

  • Page 129: Gateway Accounts (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference Line Tab FAX T38 Redundancy Select the appropriate number. The default is 1. Fax Tone Detect Mode If you want the Gateway to detect the fax tone whether the Gateway is a caller or callee, then select caller or callee.

  • Page 130: Voip Fallback To Pstn (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference Line Tab VoIP Fallback to PSTN (SPA3102/AG310) Field Description Auto PSTN Fallback If enabled, the SPA will automatically route all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down). The default is yes.

  • Page 131: Fxs Port Polarity Configuration

    Chapter 5 Linksys ATA Field Reference Line Tab Dial Plan Dial plan script for this line. The default(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) The dial plan syntax is expanded in the SPA3102 to allow the designation of three parameters to be used with a specific gateway: uid –...
  • Page 132 Chapter 5 Linksys ATA Field Reference Line Tab Linksys IP Phone Administrator Guide 5-48 Document Version 3.2...

  • Page 133: Dial Plans

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab PSTN Line Tab This section describes the fields for the following headings on the PSTN Line tab on the SPA3102 and AG310: Line Enable, page 5-33 • NAT Settings, page 5-49 •...

  • Page 134: Sip Settings

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Enter the keep alive message that should be sent periodically to maintain the current NAT Keep Alive Msg NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.
  • Page 135 Chapter 5 Linksys ATA Field Reference PSTN Line Tab SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. The default is no. EXT SIP Port The external SIP port number.
  • Page 136 Chapter 5 Linksys ATA Field Reference PSTN Line Tab SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows: none—No logging. • 1-line—Logs the start-line only for all messages. •...

  • Page 137: Proxy And Registration (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Refer-To Target Contact To contact the refer-to target, select yes. Otherwise, select no. The default is no. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE.

  • Page 138: Subscriber Information (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Proxy Fallback Intvl This parameter sets the delay (sec) after which the PAP2T will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server.

  • Page 139: Audio Configuration (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Audio Configuration (SPA3102/AG310) A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a.
  • Page 140 Chapter 5 Linksys ATA Field Reference PSTN Line Tab FAX CED Detect Enable To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no. The default is yes. G726-32 Enable To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. The default is yes.

  • Page 141: Dial Plans

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Release Unused Codec This feature allows the release of codecs not used after codec negotiation on the first call, so that other codecs can be used for the second line. To use this feature, select yes. Otherwise, select no.
  • Page 142 Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description VoIP PIN Max Retry Number of trials to allow VoIP caller to enter a PIN number (used only if authentication method is set to PIN). The default is 3. One Stage Dialing Enable one-stage dialing (applicable if authentication method is none, or HTTP Digest, or caller is in the Access List).

  • Page 143: Voip Users And Passwords (Http Authentication)

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Line 1 Fallback DP Index of the dial plan in the dial plan pool to be used when the VoIP Caller is calling from Line 1 of the same SPA3102 unit due to fallback to PSTN service when Line 1 VoIP service is down.

  • Page 144: Ring Settings

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description VoIP User 1/2/3/4/5/6/7/8 The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1 Password must therefore compute the credentials using this password, or the INVITE will be challenged with a 401 response The default is blank.
  • Page 145 Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description PSTN Answer Delay Delay in seconds before auto-answering inbound PSTN calls after the PSTN starts ringing. The range is 0-255. The default is 16. VoIP PIN Digit Timeout Timeout to wait for the 1 or subsequent PIN digits from a VoIP caller.

  • Page 146: Pstn Disconnect Detection

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description PSTN Hook Flash Len The length of the hook flash in seconds. During a PSTN-to-VoIP gateway call, the Linksys ATA processes the out-of-band hook flash signal sent from the VoIP peer through a hook-flash (momentary on-hook signal) on the FXO port.

  • Page 147: International Control (Settings)

    Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description (PSTN) Long Silence Duration This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if <Detect Long Silence> is yes. The default is 30. Silence Threshold This parameter adjusts the sensitivity of PSTN silence detection.
  • Page 148 Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Ring Validation Time Specify the minimum signal duration required by the Gateway for recognition as a ring signal. The default is 256 ms. Tip/Ring Voltage Adjust Choices are {3.1, 3.2, 3.35, 3.5}. The default is 3.5.

  • Page 149: User 1/2 Tab

    Chapter 5 Linksys ATA Field Reference User 1/2 Tab User 1/2 Tab This section describes the fields for the following headings on the User 1 and User 2 tabs: Call Forward Settings, page 5-65 • Selective Call Forward Settings, page 5-66 •...

  • Page 150: Selective Call Forward Settings

    Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description Cfwd No Ans Dest Forward number for Call Forward No Answer Service. Same as Cfwd All Dest. In addition to normal call forward destination as used in the other ATAs, on the SPA3102, you can specify the following additional parameters: •...

  • Page 151: Speed Dial Settings

    Chapter 5 Linksys ATA Field Reference User 1/2 Tab Speed Dial Settings Field Description Speed Dial 2/3/4/5/6/7/8/9 Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. The default is blank. Supplementary Service Settings The Linksys IP phone provides native support of a large set of enhanced or supplementary services.

  • Page 152: Distinctive Ring Settings

    Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description Accept Media Loopback Determines how the media loopback request is enabled. Choose automatic, never, or Request manual. The default is automatic. Media Loopback Mode Determines the media loopback mode. Choose source or mirror. Media Loopback Type Determines the media loopback type.
  • Page 153 Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description VMWI Ring Splash Len Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s). The default is .5. VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the Linksys IP phone indicating the status of the subscriber’s mail box.

  • Page 154: Pstn User Tab (Spa3102/Ag310)

    Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) PSTN User Tab (SPA3102/AG310) This section describes the fields for the following headings on the PSTN User tab on the SPA3102: • PSTN-To-VoIP Selective Call Forward Settings, page 5-70 • PSTN-To-VoIP Speed Dial Settings, page 5-70 PSTN Ring Thru Line 1 Distinctive Ring Settings, page 5-70 •...

  • Page 155: Pstn Ring Thru Line 1 Ring Settings

    Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) PSTN Ring Thru Line 1 Ring Settings Field Description Default Ring The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line 1}. If Follow Line 1 is selected, the ring to be used is determined by Line 1’s distinctive ring settings.
  • Page 156 Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) Linksys IP Phone Administrator Guide 5-72 Document Version 3.2...

  • Page 157: Appendix

    A P P E N D I X Acronyms Auto-Configuration Server Analog To Digital Converter Anonymous Call B2BUA Back to Back User Agent Bool Boolean Values. Specified as “yes” and “no”, or “1” and “0” in the profile Certificate Authority CPE Alert Signal Call Detail Record Caller ID...

  • Page 158: Appendix

    Appendix A Acronyms Foreign eXchange Station Gateway International Telecommunication Union HTML Hypertext Markup Language HTTP Hypertext Transfer Protocol HTTPS HTTP over SSL ICMP Internet Control Message Protocol IGMP Internet Group Management Protocol ILEC Incumbent Local Exchange Carrier Internet Protocol Internet Service Provider ITSP IP Telephony Service Provider Interactive Voice Response...
  • Page 159 Appendix A Acronyms Session Description Protocol SDRAM Synchronous DRAM seconds Session Initiation Protocol Shared line appearance SLIC Subscriber Line Interface Circuit Service Provider Linksys Phone Adaptor Secure Socket Layer TFTP Trivial File Transfer Protocol Transmission Control Protocol User Agent Micro-controller User Datagram Protocol Uniform Resource Locator Voicemail...
  • Page 160 Appendix A Acronyms Linksys ATA Administrator Guide Document Version 3.2...

  • Page 161: Glossary

    A P P E N D I X Glossary ACD (Automatic Call Distribution)—A switching system designed to allocate incoming calls to certain positions or agents in the order received and to hold calls not ready to be handled (often with a recorded announcement).
  • Page 162 Appendix B Glossary Dedicated access—Customers have direct access to the long-distance provider via a special circuit (T1 or private lines). The circuit is hardwired from the customer site to the POP and does not pass through the LEC switch. The dial tone is provided from the long-distance carrier. Dedicated Access Line (DAL)—Provided by the local exchange carrier.

  • Page 163: User Guidelines

    A P P E N D I X User Guidelines This appendix provides documentation for the use of the SPA products. It includes the following sections: Basic Services, page C-1 • • Enhanced Services, page C-2 The SPA can be configured to the custom requirements of the service provider, so that from the subscriber point of view, the service behaves exactly as the service provider wishes, with varying degrees of control left with the end user.

  • Page 164: Receiving A Phone Call

    Appendix C User Guidelines Enhanced Services Receiving a Phone Call Service description The SPA can receive calls from the PSTN or other IP Telephony subscribers User action required to When the telephone rings, pick up the handset and begin talking. activate or use Expected call and Each subscriber is assigned an E.164 ID (phone number) so that they may...

  • Page 165: Calling Line Identification Restriction (Clir)-Caller Id Blocking

    Appendix C User Guidelines Enhanced Services Expected call and Caller ID is sent to the distant party for this call only. Users must repeat this network behavior process at the start of each call. User action required to No action required. This service is only in effect for the duration of the deactivate or end current call.

  • Page 166: Disable Or Cancel Call Waiting

    Appendix C User Guidelines Enhanced Services Disable or Cancel Call Waiting Service description The SPA supports disabling of call waiting permanently or on a per-call basis. User action required to To temporarily disable Call Waiting (for the length of one call) do the activate or use following before placing a call: Lift Receiver...

  • Page 167: Call-Waiting With Caller Id

    Appendix C User Guidelines Enhanced Services Call-Waiting with Caller ID Service description When the user is on the phone and has Call Waiting active, the new caller Caller ID information is displayed on the user phone display screen at the same time the user is hearing the Call Waiting beeps/tones.

  • Page 168: Attendant Call Transfer

    Appendix C User Guidelines Enhanced Services Attendant Call Transfer Service description Attendant Call Transfer lets a customer use their touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones. User action required to While in a call with the party to be transferred: activate or use Press the switch hook or flash button on the phone to place the party on...

  • Page 169: Call Hold

    Appendix C User Guidelines Enhanced Services Expected call and When the user presses the switch hook or flash button, the transferee is network behavior placed on hold. When the user successfully dials the transfer number, the transferee automatically calls the dialed number. User action required to Not applicable.

  • Page 170: Three-Way Ad-Hoc Conference Calling

    Appendix C User Guidelines Enhanced Services Three-Way Ad-Hoc Conference Calling Service description This feature allows the user to conference up to two other numbers on the same line to create a three-way call. User action required to If you are already on a call and wish to add a third party: activate or use Press the switch hook or flash button Listen for dial tone...

  • Page 171: Automatic Call Back

    Appendix C User Guidelines Enhanced Services Automatic Call Back Service description This feature allows the user to place a call to the last number they tried to reach whether the call was answered, unanswered, or busy, by dialing an activation code. User action required to Pick up the receiver activate or use...

  • Page 172: Call Fwd – Busy

    Appendix C User Guidelines Enhanced Services Expected call and This feature allows a user the option to divert (forward) all calls to their network behavior telephone number to any number using the touchtone keypad of their telephone or web browser interface. This service is activated or deactivated from the phone being forwarded or the web browser interface.

  • Page 173: Call Fwd—No Answer

    Appendix C User Guidelines Enhanced Services Call FWD—No Answer Service description Calls are forwarded to the designated forwarding number after a configurable time period elapses while the SPA is ringing and does not answer. User action required to Lift the receiver activate or use Listen for dial tone Press *92Listen for dial tone and enter the telephone number you are...

  • Page 174: Distinctive/Priority Ringing And Call Waiting Tone

    Appendix C User Guidelines Enhanced Services Distinctive/Priority Ringing and Call Waiting Tone Service description The SPA supports a number of ringing and call waiting tone patterns to be played when incoming calls arrive. The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP proxy server (or other intermediate application server in the service provider domain).
  • Page 175 Appendix C User Guidelines Enhanced Services Expected call and network Pick up the receiver behavior Listen for dial tone Press single digit code assigned to the stored number (2-9) Press # to signal dialing complete The number is automatically dialed normally. User action required to None deactivate or end...
  • Page 176 Appendix C User Guidelines Enhanced Services Linksys ATA Administrator Guide C-14 Document Version 3.01...
  • Page 177 I N D E X Auth ID 5-40, 5-54 Symbols Auth Resync-Reboot 5-36, 5-51 **# command 5-71 AVT Codec Name 5-16 **1 command 5-71 AVT Dynamic Payload 5-15 Numerics 404 Forbidden 3-18 bandwidth budget 2-14 binary format Blind Attn-Xfer Enable 5-38 Blind Transfer Code 5-24...
  • Page 178 Index Call Return Code 5-24 Call Return Serv parameter 5-41 Call 1 Bytes Recv Call Type parameter Call 1 Bytes Sent Call Waiting Serv parameter 5-41 Call 1 Callback candidate sequences Call 1 Decode Latency caring for hardware Call 1 Decoder Cblk Ring Splash Len parameter 5-68 Call 1 Duration...
  • Page 179 Index multicast address 2-20 CWT Frequency 5-23 network mask 2-19 primary DNS server 2-20 static gateway IP address 2-20 Daylight Saving Time Rule WAN IP address 2-19 5-30 debugging CID_Serv parameter 3-19 5-41 CID Act Code Debug Level 5-10 5-26 Debug Level parameter CID Deact Code 5-26...
  • Page 180 Index DND Setting parameter 5-67 EXT SIP Port parameter 5-36, 5-51 DNS Query Mode DNS server check 2-20 Factory 2-20 2-21 factory defaults DNS Server Order parameter DNS SRV Auto Prefix parameter resetting 2-21 5-39, 5-53 user 2-21 Domain parameter 5-2, 5-9 FAX CED Detect Enable DTMF Playback Length...
  • Page 181 Index Hook Flash Timer Min 5-23 Hook Flash Tx Method 5-44, 5-56 G.729 voice codecs 2-14 Hook State G711a Codec Name 5-16 G711u Codec Name 5-16 Hook State parameter G723 Codec Name 5-16 Host Name parameter 5-2, 5-9 G723 Enable 5-43, 5-55 HTTP G726-16 Enable...
  • Page 182 Index MOH Server 5-38 multicast address LAN IP address check 2-20 check 2-20 2-20 Last Called Number MWI Dial Tone 5-20 Last Called PSTN Number parameter MWI Serv parameter 5-42 Last Called VoIP Number parameter Last Caller Number Last PSTN Caller parameter Last PSTN Disconnect Reason parameter NAT Keep Alive Dest 5-35, 5-50...
  • Page 183 Index PSTN Hook Flash Len parameter 5-62 PSTN Peer Name parameter parameters PSTN Peer Number parameter XML file PSTN PIN Digit Timeout parameter 5-61 Password 5-40, 5-54 PSTN PIN Tone parameter 5-20 password PSTN Ring Thru Delay parameter 5-61 2-18, 2-21 PSTN Ring Timeout parameter 5-61 pause, in dial plans...
  • Page 184 Index Remove Last Reg 5-11 RTCP Tx Interval 5-15 Reorder Delay RTP Bytes Recv parameter 5-23 Reorder Tone RTP Bytes Sent parameter 5-19 repetition, in dial plans RTP CoS Value 5-35, 5-50 resetting RTP Log Intvl parameter 5-37, 5-52 factory defaults RTP Packet Size 2-21 5-14...
  • Page 185 Index network mask 2-20 SIT3 RSC 5-14 primary DNS server SIT3 Tone 2-20 5-20 static gateway IP address SIT4 RSC 2-20 5-14 static IP addressing 2-19 SIT4 Tone 5-20 Set Local Date (mm/dd) 5-30 SPA To PSTN Gain parameter 5-63 Set Local Time (HH/mm) specifies 5-30...
  • Page 186 Index Tip/Ring Voltage Adjust parameter 5-64 VoIP Call Encoder parameter troubleshooting VoIP Caller 1/2/3/4/5/6/7/8 DP parameter 3-18 5-59 Try Backup RSC VoIP Caller 1/2/3/4/5/6/7/8 PIN parameter 5-14 5-59 VoIP Caller Authentication Method parameter 5-57 VoIP Caller ID Pattern parameter 5-59 VoIP Call FAX parameter Unattn Transfer Serv parameter 5-42...

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