Page 1 ADMINISTRATION GUIDE Cisco Small Business SPA2102, SPA3102, SPA8000, SPA8800, PAP2T Analog Telephone Adapters...
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Page 3: Table Of Contents
Contents About This Document Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices ATA Connectivity Requirements PAP2T Connectivity SPA2102 Connectivity SPA3102 Connectivity SPA8000 Connectivity SPA8800 Connectivity ATA Software Features Voice Supported Codecs SIP Proxy Redundancy Other ATA Software Features Chapter 2: Basic Administration and Configuration Basic Services and Equipment Required Downloading Firmware...
Page 4 Contents Provisioning Capabilities Configuration Profile Chapter 3: Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) NAT Mapping with Session Border Controller NAT Mapping with SIP-ALG Router Configuring NAT Mapping with a Static IP Address Configuring NAT Mapping with STUN Determining the Router’s NAT Mechanism Firewalls and SIP...
Page 5 Contents Contact List for a Trunk Group Outgoing Call Routing for a Trunk Group Configuring a Trunk Group Trunk Group Management Setting the Hunt Policy Additional Notes About Trunk Groups Chapter 5: Configuring Music on Hold Using the Internal Music Source for Music On Hold Using the Internal Music Source Changing the Music File for the Internal Music Source Configuring a Streaming Audio Server...
Page 6 Contents PSTN to VoIP Call with and Without Ring-Thru VoIP to PSTN Call With and Without Authentication Call Forwarding to PSTN Gateway (SPA3102 and SPA8800) Appendix A: ATA Routing Field Reference Router Status page Product Information section System Status section WAN Status page Internet Connection Settings section Static IP Settings section...
Page 7 Contents System Information section (PAP2T) PSTN Line Status section (SPA3102) Trunk Status section (SPA8000) System page System Configuration section Internet Connection Type section (PAP2T) Optional Network Configuration section (PAP2T) Miscellaneous Settings section (not used with PAP2T) SIP page SIP Parameters section SIP Timer Values (sec) section Response Status Code Handling section RTP Parameters section...
Page 8 Contents SIP Settings section Call Feature Settings section Proxy and Registration section Subscriber Information section Supplementary Service Subscription section Audio Configuration section Gateway Accounts section (SPA3102) VoIP Fallback to PSTN section (SPA3102 and SPA8800) Dial Plan section FXS Port Polarity Configuration section VoIP-to-PSTN Gateway Setup section (SPA8800) PSTN-To-VoIP Gateway Setup section (SPA8800) FXO Timer Values (sec) section (SPA8800)
Page 9 Contents Audio Configuration section Dial Plans section VoIP-To-PSTN Gateway Setup section VoIP Users and Passwords (HTTP Authentication) section Ring Settings section FXO (PSTN) Timer Values (sec) section PSTN Disconnect Detection section International Control (Settings) section User page Call Forward Settings section Selective Call Forward Settings section Speed Dial Settings section Supplementary Service Settings section...
Page 10 Contents Product Resources Related Documentation Cisco Small Business ATA Administration Guide...
Page 11: About This Document
Preface About This Document This guide is intended to help VARs and Service Providers to manage and configure the Cisco Analog Telephone Adapters (ATAs). This preface provides helpful information about this guide and other resources that are available to you. Before you begin to use this guide, refer to the following topics: •...
Page 12: Document Conventions
Preface Document Conventions The following are the typographic conventions used in this document. Typographic Meaning Element Boldface May indicate either of the following: • A user interface element that you need to click, select, or otherwise act on • A literal value to be entered in a field. Italic May indicate either of the following: •...
Page 13: Finding Text In A Pdf
Preface Finding Information in PDF Files The SPA9000 Voice System documents are published as PDF files. The PDF Find/ Search tool within Adobe® Reader® lets you find information quickly and easily online. You can perform the following tasks: • Search an individual PDF file. •...
Page 14 Preface Finding Text in Multiple PDF Files Search window lets you search for terms in multiple PDF files that are stored on your PC or local network. The PDF files do not need to be open. Start Acrobat Professional or Adobe Reader. STEP 1 Find Choose Edit >...
Page 15 Preface When the Results appear, click + to open a folder, and then click any link to open STEP 4 the file where the search terms appear. For more information about the Find and Search functions, see the Adobe Acrobat online help.
Page 16: Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters
SPA9000 Voice System or legacy PBX, or with an Internet-based call-control system. Figure 1 ATA Deployment without Onsite Call Control Voice Layer 3 gateway IP infrastructure Broadband PSTN Telephone/fax Ethernet Linksys ATA Broadband CPE (DSL, cable, fixed wireless) SIP proxy Cisco Small Business ATA Administration Guide...
Page 17: Comparison Of Ata Devices
Layer 3 gateway IP infrastructure Telephone/ Broadband PSTN Ethernet Linksys ATA Broadband CPE (DSL, cable, fixed wireless) SIP proxy This chapter introduces the functionality of the ATA devices and describes the features that are available. Refer to the following topics: •...
Page 18 Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices ATA Models Product RJ-45 RJ-45 Voice Description Name (Analog PSTN Internet Ethernet Lines Phone) (WAN) (LAN) PAP2T — — Voice adapter with two FXS ports. SPA2102 — Voice adapter with router. SPA3102 Voice adapter with router and PSTN connectivity.
Page 19 Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Figure 3 How ATAs Provide Voice Connectivity Ethernet/Wireless WRP400, RTP300, WRTP54G, and SPA2102 Fax (up to 4 SPA8000) PSTN DSL/cable modem Broadband WAG54GP2, router AG310 Internet SPA8000, Broadband PAP2T router Analog phone (up to 8 with...
Page 20: Ata Connectivity Requirements
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements ATA Connectivity Requirements An ATA device can be connected to a local router, or directly to the Internet. Each phone connected to an RJ-11 (analog) port on the ATA device connects to other devices through SIP, which is transmitted over the IP network.
Page 21: Pap2T Connectivity
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements PAP2T Connectivity As shown in the following figure, the PAP2T has two FXS ports (voice lines 1 and Administrative IVR (Line 1 or IP Router (with Line 2) hairpinning) or Broadband modem Ethernet port...
Page 22: Spa2102 Connectivity
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements SPA2102 Connectivity As shown in the following illustration, the SPA2102 has two FXS ports (voice lines 1 and 2). Administrative IVR (Line 1 or IP Router (with Line 2) hairpinning) or Broadband modem Ethernet port...
Page 23: Spa3102 Connectivity
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements SPA3102 Connectivity As shown in the following figure, the SPA3102 has one FXS port (voice line 1). Administrative IVR (Line 1 or IP Router (with Line 2) hairpinning) or Broadband modem Ethernet port Line 1...
Page 24: Spa8000 Connectivity
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements SPA8000 Connectivity As shown in the following illustration, the SPA8000 consists of eight voice ports (voice lines 1-8). 8 FXS (RJ-11/RJ-21 ) ports Administrative IVR (Line 1 or IP Router (with SPA8000 Line 2) hairpinning) or...
Page 25: Spa8800 Connectivity
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements NOTE • You can use line 1 or line 2 to access the IVR functions. • 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.
Page 26: Ata Software Features
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features By default, the device on the LAN port is assigned the network address 192. 1 68.0.0 with a subnet mask of 255.255.255.0. If there is a network address conflict with a device on the Ethernet port, the network address of the device on the LAN port is automatically changed to 192.
Page 27 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features You can configure your preferred codec in Configuration Utility. See “SDP Payload Types section,” on page 134 “Audio Configuration section,” on page 174. See also “Supported Codecs,” on page 54 for a list of which codecs are supported on each ATA device.
Page 28: Sip Proxy Redundancy
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 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 thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance.
Page 29 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description • Modem pass-through mode can be triggered only by Modem and Fax Modem Line Toggle Code. predialing the number set in the Pass-Through (Set in the Regional tab.) •...
Page 30 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description Secure Calls A user (if enabled by service provider or administrator) has the option to make an outbound call secure in the sense that the audio packets in both directions are encrypted.
Page 31 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description Signaling Hook The ATA device can signal hook flash events to the remote Flash Event party on a connected call. This feature can be used to provide advanced mid-call services with third-party-call- control.
Page 32 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description Calling Party Calling Party Control (CPC) signals to the called party Control equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily.
Page 33 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description SIP Over TLS SPA2102, SPA3102, and SPA8800 devices allow the use of SIP over Transport Layer Security (TLS). SIP over TLS is designed to eliminate the possibility of malicious activity by encrypting the SIP messages of the service provider and the end user.
Page 34 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description Register Retry The Register Retry Enhancements feature for SPA2102, Enhancements SPA3102, and PAP2T devices adds flexibility to the delay timers that are activated when the SIP REGISTER of a device fails.
Page 35 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description DHCP Renewal on SPA2102, SPA3102, and PAP2T voice devices typically Timeout operate in a network where a DHCP server assigns IP addresses to the devices. Because IP addresses are a limited resource, the DHCP server periodically renews the device lease on the IP address.
Page 36: Chapter 2: Basic Administration And Configuration
Basic Administration and Configuration This chapter describes the equipment and services that are required to install your ATA device and explains how to complete the basic administration and configuration tasks. Refer to the following topics: • “Basic Services and Equipment Required” section on page 36 •...
Page 37: Downloading Firmware
Basic Administration and Configuration Downloading Firmware • Computer with Microsoft Windows XP or Windows Vista (for system configuration) • Analog phones • UPS (uninterruptible Power Source) recommended for devices such as the Integrated Access Device, network switch, router, and PoE switch to ensure that your phone system continues to work during a power failure, just like your home phone.
Page 38: Setting Up Your Ata Device
Basic Administration and Configuration Setting up Your ATA Device Make a note of the IP address that is announced. STEP 2 If the administration computer is connected to the Ethernet port of the ATA NOTE device, the default IP address is 192. 1 68.0. 1 . Use the administration computer to install the latest firmware: STEP 3 a.
Page 39: Using The Administration Web Server
Basic Administration and Configuration Using the Administration Web Server By default, there are no passwords assigned for either the Administrator account or NOTE the User account. The Administrator account can modify all the web profile parameters and the passwords of both Administrator and User account. The User account can access only part of the web profile parameters.
Page 40: Connecting To The Administration Web Server
Basic Administration and Configuration Using the Administration Web Server • “Advanced Configurations” section on page 43 Connecting to the Administration Web Server To access the ATA administration web server, perform the following steps. Start Internet Explorer on a computer that is connected to the same network as the STEP 1 ATA device.
Page 41 Basic Administration and Configuration Using the Administration Web Server Complete the WAN configuration for DHCP, static IP addressing, or PPPoE. STEP 3 For DHCP: Connection Type a. Select DHCP from the drop-down menu. b. If you use a cable modem, you may need to configure the MAC Clone Settings. (Contact your ISP for more information.) c.
Page 42: Registering To The Service Provider
Basic Administration and Configuration Using the Administration Web Server Registering to the Service Provider To use VoIP phone service, you must configure your ATA device to the Service Provider. Start Internet Explorer, connect to the administration web server, and choose STEP 1 Admin access with Advanced settings.
Page 43: Advanced Configurations
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device Advanced Configurations Other parameters may need to be changed from the defaults, depending on the requirements of a specific ITSP. Some of the commonly configured parameters include the following: •...
Page 44: Resync Url
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device protocol server-name If no is specified, TFTP is assumed. If no is specified, the server-name host that requests the URL is used as If no port specified, the default port of the protocol is used. (69 for TFTP or 80 for HTTP) firmware-pathname is typically the file name of the binary located in a...
Page 45: Provisioning Your Ata Device
Basic Administration and Configuration Provisioning Your ATA Device Provisioning Your ATA Device This section describes the provisioning functionality of the ATA device. This section includes the following topics: • “Provisioning Capabilities” section on page 45 • “Configuration Profile” section on page 46 For detailed information about provisioning your ATA device, refer to the SPA Provisioning Guide.
Page 46: Configuration Profile
Basic Administration and Configuration Provisioning Your ATA Device Configuration Profile The 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>...
Page 47: Chapter 3: Configuring Your System For Itsp Interoperability
Configuring Your System for ITSP Interoperability This chapter provides configuration details to help you to ensure that your infrastructure properly supports voice services. • “Network Address Translation (NAT) and Voice over IP (VoIP),” on page 47 • “Firewalls and SIP,” on page 53 •...
Page 48: Nat Mapping With Session Border Controller
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) NAT Mapping with Session Border Controller It is strongly recommended that you choose an ITSP that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the ITSP, you have more choices in selecting a router.
Page 49 Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Scroll down to the NAT Support Parameters section, and then enter the following STEP 3 settings to support static mapping to your public IP address: •...
Page 50: Configuring Nat Mapping With Stun
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Configuring NAT Mapping with STUN If the ITSP network does not provide a Session Border Controller functionality, and if other requirements are met, it is possible to use STUN as a mechanism to discover the NAT mapping.
Page 51 Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Voice tab > SIP > NAT Support Parameters Click Voice tab > Line , where N is the number of the line interface. STEP 4 Scroll down to the NAT Settings section.
Page 52: Determining The Router's Nat Mechanism
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Determining the Router’s NAT Mechanism STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port.
Page 53: Firewalls And Sip
Configuring Your System for ITSP Interoperability Firewalls and SIP Click Submit All Changes. STEP 6 View the syslog messages to determine whether your network uses symmetric STEP 7 NAT. Look for a warning header in the REGISTER messages, such as Warning: 399 spa "Full Cone NAT Detected.”...
Page 54: Chapter 4: Configuring Voice Services
Configuring Voice Services This chapter describes how to configure your ATA device to meet the customer’s requirements for voice services. • “Supported Codecs,” on page 54 • “Using a FAX Machine,” on page 55 • “Managing Caller ID Service,” on page 58 •...
Page 55: Using A Fax Machine
Configuring Voice Services Using a FAX Machine • G.726-40 • G.729a • G.723 WRTP54G • G.711u (configured by default) • G.711a • G.726-32 • G.729a • G.723 Using a FAX Machine You can connect a fax machine to an FXS port on the SPA2102, SPA3102, SPA8000, and SPA8800.
Page 56: Fax Troubleshooting
Configuring Voice Services Using a FAX Machine • Echo Canceller: no • Silence suppression: no • Preferred Codec: G.711 • Use pref. codec only: yes If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax STEP 4 relay) and enable fax using modem passthrough.
Page 57 Configuring Voice Services Using a FAX Machine Monitor the network and record the following statistics: STEP 4 • Jitter • Loss • Delay If faxes fail consistently, capture a copy of the web interface settings by selecting STEP 5 Save As > Web page, complete from the administration web server page. You can send this configuration file to Technical Support.
Page 58: Managing Caller Id Service
Configuring Voice Services Managing Caller ID Service Managing Caller ID Service The choice of caller ID (CID) method is dependent on your area/region. To configure CID, use the following parameters: Parameter Description and Value Caller ID Regional The following choices are available: Method •...
Page 59 Configuring Voice Services Managing Caller ID Service There are three types of Caller ID: • 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 the following figure (a) –...
Page 60: Silence Suppression And Comfort Noise Generation
Configuring Voice Services Silence Suppression and Comfort Noise Generation Silence Suppression and Comfort Noise Generation Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls supported by the network by reducing the required bandwidth for a single call.
Page 61: Configuring Dial Plans
Configuring Voice Services Configuring Dial Plans Configuring Dial Plans Dial plans determine how the digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international.
Page 62: Digit Sequences
Configuring Voice Services Configuring Dial Plans Digit Sequence Function Enter any of these characters to represent a key 0 1 2 3 4 5 6 7 8 9 0 that the user must press on the phone keypad. Enter x to represent any character on the phone keypad.
Page 63 Configuring Voice Services Configuring Dial Plans Digit Sequence Function For an intersequence tone, enter a comma (comma) between digits to play an “outside line” dial tone after a user-entered sequence. EXAMPLE: 9, 1xxxxxxxxxx An “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1.
Page 64: Digit Sequence Examples
Configuring Voice Services Configuring Dial Plans Digit Sequence Examples The following examples show digit sequences that you can enter in a dial plan. In a complete dial plan entry, sequences are separated by a pipe character (| ) , and the entire set of sequences is enclosed within parentheses.
Page 65 Configuring Voice Services Configuring Dial Plans This is example is useful where a local area code is required 8, <:1212>xxxxxxx by the carrier but the majority of calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number.
Page 66: Acceptance And Transmission The Dialed Digits
Configuring Voice Services Configuring Dial Plans Acceptance and Transmission the Dialed Digits 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.
Page 67: Dial Plan Timer (Off-Hook Timer)
Configuring Voice Services Configuring Dial Plans Dial Plan Timer (Off-Hook Timer) You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts counting when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated.
Page 68: Interdigit Long Timer (Incomplete Entry Timer)
Configuring Voice Services Configuring Dial Plans Interdigit Long Timer (Incomplete Entry Timer) You can think of this timer as the “incomplete entry” timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan.
Page 69 Configuring Voice Services Configuring Dial Plans Syntax for the Interdigit Short Timer • SYNTAX 1: S:s, ( dial plan ) Use this syntax to apply the new setting to the entire dial plan within the parentheses. • SYNTAX 2: sequence Ss Use this syntax to apply the new setting to a particular dialing sequence.
Page 70: Editing Dial Plans
Configuring Voice Services Configuring Dial Plans Editing Dial Plans You can edit dial plans and can modify the control timers. Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin STEP 1 Login and then click Advanced. Entering the Line Interface Dial Plan This dial plan is used to strip steering digits from a dialed number before it is transmitted out to the carrier.
Page 71: Secure Call Implementation
Configuring Voice Services Secure Call Implementation Secure Call Implementation This section describes secure call implementation with the ATA device . It includes the following topics: • “Enabling Secure Calls” section on page 71 • “Secure Call Details” section on page 72 •...
Page 72: Secure Call Details
Configuring Voice Services Secure Call Implementation The ATA device 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.
Page 73: Using A Mini-Certificate
Configuring Voice Services Secure Call Implementation Using a Mini-Certificate 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).
Page 74: Configuring Voice Services
Configuring Voice Services Secure Call Implementation The partner sites require a logon. NOTE The Mini Certificate Generator uses the following syntax: ca-key user-name user-id expire-date gen_mc 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: 9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qq e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ...
Page 75: Sip Trunking And Hunt Groups On The Spa8000
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 SIP Trunking and Hunt Groups on the SPA8000 The SPA8000 supports SIP Trunking, which allows you to connect a traditional PBX to VoIP services. In this configuration, calls go through the ITSP rather than the PSTN, yet the call routing functionality is similar to that of traditional PSTN lines.
Page 76: About Sip Trunking
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 About SIP Trunking The SIP Trunking feature allows a traditional PBX to seamlessly migrate from PSTN service to VoIP service over a broadband link. The SPA8000 offers up to eight telephone lines to the PBX.
Page 77 Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 The following figure shows a simplified logical block diagram of the SPA8000 with the SIP Trunking feature. Figure 1 Logical Block Diagram of SIP Trunking Phone 1 Phone 2 Phone 3 Phone 4 Phone 5...
Page 78: Setting The Trunk Group Call Capacity
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Although the figure shows only one ITSP account, each standalone line and each NOTE Trunk Group can be configured with a different ITSP (with some limitations applied). Setting the Trunk Group Call Capacity The ITSP may set a limit to the number of calls that can be made on a trunk group.
Page 79: Contact List For A Trunk Group
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 If the call is picked up by the PBX, the Line UA replies 200 OK with SDP to the STEP 5 internal Proxy. The Trunk UA in turn replies 200 OK to the ITSP and relay the Line SDP in the 200 OK message also.
Page 80 Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 below), the hunt proceeds randomly through the unchosen lines until each line is tried. al: All. The Trunk UA rings all the lines at the same time. • interval: The number of seconds to wait for one line to answer, before choosing another line.
Page 81: Outgoing Call Routing For A Trunk Group
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 chooses lines in random order (hunt=ra). If a selected line does not answer within 12 seconds (12), the Trunk UA chooses another line at random. If there is no answer after 1 cycle (1), the call is forwarded to forwarded to the specified number (cfwd=14085550123).
Page 82: Configuring A Trunk Group
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Configuring a Trunk Group To configure a hunt group, you must first specify the trunk lines by assigning lines to trunk groups. Then you enter the account information, specify the call capacity, and configure the Contact List.
Page 83: Trunk Group Management
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Enter the settings for each trunk group, as needed: STEP 3 a. Click Voice tab > T , where n represents the trunk group number (T1 ... T4). b. Enter the account information in the Subscriber Information section. •...
Page 84 Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Trunk Status page The Trunk Status page shows all calls that are currently active on each trunk group. This page shows a snapshot of the trunk activity. You can refresh the data at any time by clicking the Refresh button on the web browser toolbar.
Page 85: Setting The Hunt Policy
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Setting the Hunt Policy You can configure the SPA8000 so that the hunt rule applies to all phone or only to the phones that are on hook. Connect to the administration web server, and choose Admin access with STEP 1 Advanced settings.
Page 86: Chapter 5: Configuring Music On Hold
Configuring Music on Hold This chapter explains how to configure Music on Hold using either a music file or streaming audio. This chapter includes the following topics: • “Using the Internal Music Source for Music On Hold,” on page 86 •...
Page 87: Changing The Music File For The Internal Music Source
Configuring Music on Hold Using the Internal Music Source for Music On Hold Start Internet Explorer, and then enter the IP address of the telephone. The STEP 2 Telephone Configuration page appears in a separate browser window. Click Admin Login, and then click Advanced. STEP 3 Click the Ext 1 tab.
Page 88: Configuring A Streaming Audio Server
Configuring Music on Hold Configuring a Streaming Audio Server • path: The location and name of a music file in the correct format • For example, if the computer local IP address is 192. 1 68.0.5, the directory is musicdir jazzmusic.dat named , and the converted music file is named...
Page 89 Configuring Music on Hold Configuring a Streaming Audio Server After you complete the required configuration, the FXS port is ready to stream audio. The functionality depends on the hook state of the FXS port: • If the FXS port is off hook, an incoming call is answered automatically and audio is streamed to the calling party.
Page 90: Configuring The Streaming Audio Server
Configuring Music on Hold Configuring a Streaming Audio Server Configuring the Streaming Audio Server Use the following procedure to configure an SAS with an external music source. Connect an RJ-11 adapter between the music source (a CD player or iPod, for STEP 1 example) and an FXS port.
Page 91: Using The Ivr With An Sas Line
Configuring Music on Hold Configuring a Streaming Audio Server g. Close the window for the Telephone Configuration page. h. Repeat this step to configure each phone, as needed. 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: Power off the ATA device.
Page 92: Chapter 6: Configuring The Pstn (Fxo) Gateway On The Spa3102
Configuring the PSTN (FXO) Gateway on the SPA3102 This chapter describes how to configure the PSTN gateway on the SPA3102 and the SPA8800. • “Connecting to PSTN and VoIP Services” section on page 92 • “How VoIP-To-PSTN Calls Work” section on page 93 •...
Page 93: How Voip-To-Pstn Calls Work
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work FXO Port: The SPA3102 has 1 FXO port that you can connect to the PSTN. Configure the FXO settings by using the SPA3102 PSTN Line page. • The SPA8800 is designed to work with your PBX as a PSTN gateway and a VoIP gateway.
Page 94: Two-Stage Dialing (Spa3102)
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work Two-Stage Dialing (SPA3102) 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 User ID or with a user-id that matches exactly the <...
Page 95: How Pstn-To-Voip Calls Work
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work When the source address of the INVITE is 127.0.0. 1 , authentication is automatically NOTE disabled because this is a call by the local user. This applies to both one-stage and two-stage dialing.
Page 96: Terminating Gateway Calls
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work For information about configuring the timer values for the above scenarios, see NOTE “FXO (PSTN) Timer Values (sec) section,” on page 214. For information about configuring caller authentication on the SPA3102, see “VoIP- To-PSTN Gateway Setup section,”...
Page 97: Voip Outbound Call Routing (Spa3102)
Configuring the PSTN (FXO) Gateway on the SPA3102 VoIP Outbound Call Routing (SPA3102) VoIP Outbound Call Routing (SPA3102) On the SPA3102, 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 that the user enters.
Page 98: Configuring Voip Failover To Pstn
Configuring the PSTN (FXO) Gateway on the SPA3102 Configuring VoIP Failover to PSTN You can set up multiple PSTN gateways at different locations and configure Line 1 to use a different gateway when dialing specific numbers. Configuring VoIP Failover to PSTN When power is disconnected from the ATA, the FXS port is connected to the FXO port.
Page 99: Other Options
Configuring the PSTN (FXO) Gateway on the SPA3102 Other Options • 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.
Page 100: Call Progress Tones (Spa3102 And Spa8800)
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios Call Progress Tones (SPA3102 and SPA8800) The SPA3102 and the SPA8800 have configurable call progress tones. Call progress tones are generated locally on the ATA, so an end user is advised of status (such as ringback).
Page 101: Pstn To Voip Call With And Without Ring-Thru
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios PSTN to VoIP Call with and Without Ring-Thru The PSTN caller calls the PSTN line connected to the FXO port. Ring-Thru is PSTN Answer Delay disabled. After the call rings for a delay equal to the value in the VoIP gateway answers the call and prompts the PSTN caller to enter a PIN number (assuming PIN authentication is enabled).
Page 102 Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 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.
Page 103: Call Forwarding To Pstn Gateway (Spa3102 And Spa8800)
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios HTTP Digest Authentication is one way to perform one-stage dialing of a VoIP-To- NOTE 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.
Page 104 Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios If the PSTN Line is busy at the moment of the forward, it does not answer the VoIP NOTE call. The call forward rule is ignored and Line 1 continues to ring. Forward-All to the PSTN gateway Cfwd All Dest In this scenario, Line 1 is configured with...
Page 105: Appendix A: Ata Routing Field Reference
ATA Routing Field Reference This chapter describes the settings that you can configure under the Router and Network tabs in the administration web server pages. This information applies to the SPA2102, SPA3102, SPA8000, and SPA8800 NOTE models. To configure router settings for the PAP2T, and WRTP54G, see the user guide for the router.
Page 106: Product Information Section
ATA Routing Field Reference Router Status page Router tab > Status page > Product Information section Product Name Model number of the ATA device. Serial Number Serial number of the ATA device. Software Version Version number of the ATA software. Hardware Version Version number of the ATA hardware.
Page 107: Wan Status Page
ATA Routing Field Reference WAN Status page Current Time Current date and time of the system; for example, 10/3/ 2003 16:43:00. 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.
Page 108: Static Ip Settings Section
ATA Routing Field Reference WAN Status page Router tab > WAN Setup page > Static IP Settings section Static IP Static IP address of ATA device, which takes effect if DHCP is disabled. The default is 0.0.0.0. NetMask The NetMask used by ATA device when DHCP is disabled.
Page 109: Optional Settings Section
ATA Routing Field Reference WAN Status page Router tab > WAN Setup page > Optional Settings section HostName The host name of the ATA device. Domain The network domain of the ATA device. Primary DNS The DNS server that is used by the ATA device. NOTE: When DHCP is enabled, you can enter the IP address of a DNS server in addition to DHCP-supplied...
Page 110: Mac Clone Settings Section
ATA Routing Field Reference WAN Status page Router tab > WAN Setup page > MAC Clone Settings section A MAC address is a 12-digit code assigned to a unique piece of hardware for identification, like a social security number. Some ISPs require you to register a MAC address in order to access the Internet.
Page 111: Vlan Settings Section
ATA Routing Field Reference LAN Status page Maximum Uplink The maximum bandwidth for LAN to WAN throughput. Speed The default is 128 kbps. Router tab > WAN Setup page > VLAN Settings section Enable VLAN Allows (yes) or prevents (no) VLAN access. NOTE: Choose yes if your ATA device is connected to a switch that uses VLAN tagging.
Page 112: Lan Networking Settings Section
ATA Routing Field Reference Application page Router tab > LAN Setup page > LAN Networking Settings section Use these network settings when using NAT. LAN IP Address IP address of the ATA device on the LAN side. LAN Subnet Mask IP address for subnet mask.
Page 113: Port Forwarding Settings Section
ATA Routing Field Reference Application page • “System Reserved Ports Range section,” on page 114 Router tab > Application page > Port Forwarding Settings section This feature allows you to set up specialized Internet applications that require port forwarding on a range of ports. Enable Options are Yes or No.
Page 114: Miscellaneous Settings Section
ATA Routing Field Reference Application page Router tab > Application page > Miscellaneous Settings section Multicast Passthru Used for passing multicast traffic. Options are disabled, inbound, outbound, inbound and outbound. Router tab > Application page > System Reserved Ports Range section Starting Port A port identified as a reserve port and that is not used for NAT translation.
Page 115: Appendix B: Ata Voice Field Reference
ATA Voice Field Reference This chapter describes the settings that you can configure under the Voice tab in the administration web server pages. For information about the Voice > Provisioning tab, see the SPA Provisioning NOTE Guide. After you click the Voice tab, you can choose the following pages: •...
Page 116: Info Page
ATA Voice Field Reference Info page Info page You can use the Voice tab > Info page to view information about the ATA device. With some variations, depending on the model, this page includes the following sections: • “Product Information section,” on page 116 •...
Page 117: System Status Section
ATA Voice Field Reference Info page Voice tab > Info page > System Status section 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.
Page 118 ATA Voice Field Reference Info page Message Waiting Indicates whether you have new voice mail waiting. Options are either Yes or No. The value automatically is set to Yes when a message is received. You also can clear or set the flag manually. Setting this value to Yes can activate stutter tone and VMWI signal.
Page 119 ATA Voice Field Reference Info page Call 1 and 2 Indicates whether the far end has placed the call on hold. Remote Hold Call 1 and 2 Indicates whether the call was triggered by a call back Callback request. Call 1 and 2 Peer Name of the internal phone.
Page 120: System Information Section (Pap2T)
ATA Voice Field Reference Info page Voice tab > Info page > System Information section (PAP2T) DHCP Indicates if DHCP is enabled. Current IP Displays the current IP address assigned to the ATA device. Host Name Displays the current IP address assigned to the ATA device. Domain Displays the network domain name of the ATA device.
Page 121 ATA Voice Field Reference Info page Last PSTN Reason for SPA hanging up the FXO port. Can be one of the Disconnect Reason following: • PSTN Disconnect Tone • PSTN Activity Timeout • CPC Signal • Polarity Reversal • VoIP Call Failed •...
Page 122 ATA Voice Field Reference Info page VoIP State May take one of the following values: • Idle • Collecting PSTN Pin • Invalid PSTN PIN • PSTN Caller Accepted • Connected to PSTN PSTN State May take one of the following values: •...
Page 123: Trunk Status Section (Spa8000)
ATA Voice Field Reference Info page VoIP Call Decode Number of milliseconds for decoder latency. Latency VoIP Call Jitter Number of milliseconds for receiver jitter. VoIP Call Round Number of milliseconds for delay. Trip Delay VoIP Call Packets Number of packets lost. Lost VoIP Call Packet Number of invalid packets received.
Page 124: System Page
ATA Voice Field Reference System page System page You can use the Voice tab > System page to configure your system and network connections. With some variations, depending on the model, this page includes the following sections: • “System Configuration section” section on page 124 •...
Page 125: Internet Connection Type Section (Pap2T)
ATA Voice Field Reference System page Voice tab > System page > Internet Connection Type section (PAP2T) DHCP Enable or disable DHCP. The default is yes. Static IP Static IP address of ATA device, which takes effect if DHCP is disabled. The default is 0.0.0.0.
Page 126: Miscellaneous Settings Section (Not Used With Pap2T)
ATA Voice Field Reference System page DNS Server Order Specifies the method for selecting the DNS server. The options are Manual (enter the IP address of the DNS server manually; that is do not look at the DHCP-supplied DNS table), Manual/DHCP, and DHCP/Manual. DNS Query Mode Do parallel or sequential DNS Query.
Page 127: Sip Page
ATA Voice Field Reference SIP page 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.
Page 128 ATA Voice Field Reference SIP page SIP User Agent User-Agent header used in outbound requests. Name 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.
Page 129: Sip Timer Values (Sec) Section
ATA Voice Field Reference SIP page Escape Display Lets you keep the Display Name private. Select yes if you Name want the ATA device to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages.
Page 130 ATA Voice Field Reference SIP page 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. SIP Timer H INVITE final response, time-out value, which can range from 0 to 64 seconds.
Page 131: Response Status Code Handling Section
ATA Voice Field Reference SIP page Reg Retry Long When registration fails with a SIP response code that does Intvl not match Retry Reg RSC , the ATA device waits for the specified length of time before retrying. If this interval is 0, the ATA device stops trying.
Page 132: Rtp Parameters Section
ATA Voice Field Reference SIP page 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 ATA device retries registration after failing during the last registration.
Page 133 ATA Voice Field Reference SIP page 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 ATA device 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).
Page 134: Sdp Payload Types Section
ATA Voice Field Reference SIP page Voice tab > SIP page > SDP Payload Types section NSE Dynamic NSE dynamic payload type. The valid range is 96-127. Payload The default is 100. AVT Dynamic AVT dynamic payload type. The valid range is 96-127. Payload The default is 101.
Page 135: Nat Support Parameters Section
ATA Voice Field Reference SIP page G726r32 Codec G.726-32 codec name used in SDP. Name The default is G726-32. G726r40 Codec G.726-40 codec name used in SDP. Name The default is G726-40. G729a Codec G.729a codec name used in SDP. Name The default is G729a.
Page 136 ATA Voice Field Reference SIP page Insert VIA rport Inserts the parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. The default is no. Substitute VIA Addr Lets you use NAT-mapped IP:port values in the VIA header.
Page 137 ATA Voice Field Reference SIP page EXT RTP Port Min External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range. The default is 0.
Page 138: Trunking Parameters Section (Spa8000)
ATA Voice Field Reference SIP page Voice tab > SIP page > Trunking Parameters section (SPA8000) The trunking parameters apply to the Trunk Groups that you configure on the Trunk Group pages. SIP Trunking is available on the SPA8000 only. Proxy Debug This feature controls which proxy debuy messages to log.
Page 139: Regional Page
ATA Voice Field Reference Regional page Hunt Policy This parameter can be used to modify the hunting behavior for trunk lines, based on the call state of the trunk lines that are specified in the Voice tab > Trunk page, Contact List field.
Page 140: Call Progress Tones Section
ATA Voice Field Reference Regional page Voice tab > Regional page > Call Progress Tones section Dial Tone Prompts the user to enter a phone number. Reorder Tone is Dial Tone played automatically when or any of its alternatives times out. The default is 350@-19,440@-19;10(*/0/1+2).
Page 141 ATA Voice Field Reference Regional page Ring Ring Back 2 Tone Your ATA device plays this ringback tone instead of Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request. The default value is the same as Ring Back Tone , except the...
Page 142: Distinctive Ring Patterns Section
ATA Voice Field Reference Regional page Holding Tone Informs the local caller that the far end has placed the call on hold. The default is 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1). Conference Tone Played to all parties when a three-way conference call is in progress. The default is 350@-19;20(.1/.1/1,.1/9.7/1).
Page 143: Distinctive Call Waiting Tone Patterns Section
ATA Voice Field Reference Regional page Ring3 Cadence Cadence script for distinctive ring 3. The default is 60(.8/.4,.8/4). Ring4 Cadence Cadence script for distinctive ring 4. The default is 60(.4/.2,.3/.2,.8/4). Ring5 Cadence Cadence script for distinctive ring 5. The default is 60(.2/.2,.2/.2,.2/.2,1/4). Ring6 Cadence Cadence script for distinctive ring 6.
Page 144: Distinctive Ring/Cwt Pattern Names Section
ATA Voice Field Reference Regional page CWT6 Cadence Cadence script for distinctive CWT 6. The default is 30(.3/.1,.3/.1,.1/9.1). 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 145: Ring And Call Waiting Tone Spec Section
ATA Voice Field Reference Regional page Ring7 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call. The default is Bellcore-r7. Ring8 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call. The default is Bellcore-r8.
Page 146: Control Timer Values (Sec) Section
The default is 440@-10. Synchronized Ring If this is set to Yes, when a device calls the Linksys ATA, both lines ring at the same time (similar to a regular PSTN line). After one line answers, the other stops ringing. This field is only found in the PAP2T.
Page 147 ATA Voice Field Reference Regional page Call Back Delay Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA device still considers the call as failed and keeps on retrying. The default is 0.5.
Page 148: Vertical Service Activation Codes Section
ATA Voice Field Reference Regional page CPC Duration Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0.
Page 149 ATA Voice Field Reference Regional page Cfwd All Deact Cancels call forwarding of all calls. Code The default is *73. Cfwd Busy Act Forwards busy calls to the extension specified after the Code activation code. The default is *90. Cfwd Busy Deact Cancels call forwarding of busy calls.
Page 150 ATA Voice Field Reference Regional page CW Per Call Act Enables call waiting for the next call. Code The default is *71. CW Per Call Deact Disables call waiting for the next call. Code The default is *70. Block CID Act Code Blocks caller ID on all outbound calls. The default is *67.
Page 151 ATA Voice Field Reference Regional page Dist Ring Act Code Enables the distinctive ringing feature. The default is *26 Dist Ring Deact Disables the distinctive ringing feature. Code The default is *46. Speed Dial Act Assigns a speed dial number. Code The default is *74.
Page 152 ATA Voice Field Reference Regional page Referral Services These codes tell the ATA device what to do when the user Codes places the current call on hold and is listening to the second dial tone. One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, etc.
Page 153 ATA Voice Field Reference Regional page Feature Dial These codes tell the ATA device what to do when the user Services Codes 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 154: Vertical Service Announcement Codes Section (Spa2102, Spa8000)
ATA Voice Field Reference Regional page Voice tab > Regional page > Vertical Service Announcement Codes section (SPA2102, SPA8000) Service Annc Base Base number for service announcements. Number Service Annc Extension codes for service announcements. Extension Codes Voice tab > Regional page > Outbound Call Codec Selection Codes section These codes automatically appended to the dial-plan.
Page 155 ATA Voice Field Reference Regional page Force G723 Code Makes this codec the only codec that can be used for the associated call. The default is *02723. Prefer G726r16 Makes this codec the preferred codec for the associated Code call. The default is *0172616.
Page 156: Miscellaneous Section
ATA Voice Field Reference Regional page Voice tab > Regional page > Miscellaneous section Set Local Date Sets the local date (mm stands for months and dd stands (mm/dd) for days). The year is optional and uses two or four digits. Set Local Time (HH/ Sets the local time (hh stands for hours and mm stands for minutes).
Page 157 ATA Voice Field Reference Regional page Daylight Saving Enter the rule for calculating daylight saving time; it should Time Rule include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below.
Page 158 ATA Voice Field Reference Regional page Daylight Saving Daylight Saving Time can be turned on or off. This option Time Enable affects the time stamp on CallerID and affects all the lines and extensions of the phone. Default is Yes (on). FXS Port Input Gain Input gain in dB, up to three decimal places.
Page 159 ATA Voice Field Reference Regional page Caller ID Method The following choices are available: • Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS). • DTMF (Finland, Sweden)—CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.
Page 160: Line Page
ATA Voice Field Reference Line page More Echo Enable or disable more echo suppresion. The default is no. Suppression This field is not found in the PAP2T. Line page Depending on the ATA device, there may be one or more Line pages (L 1, L2, and so on).
Page 161: Line Enable Section
ATA Voice Field Reference Line page • “Gateway Accounts section (SPA3102)” section on page 180 • “Dial Plan section” section on page 181 • “FXS Port Polarity Configuration section” section on page 183 • “VoIP-to-PSTN Gateway Setup section (SPA8800),” on page 183 •...
Page 162: Streaming Audio Server (Sas) Section
ATA Voice Field Reference Line page Voice tab > Line page > Streaming Audio Server (SAS) section On the SPA8800, these settings are configured on the Phone pages only. SAS Enable To enable the use of the line as a streaming audio source, select yes.
Page 163: Nat Settings Section
ATA Voice Field Reference Line page SAS Inbound RTP This setting works around devices that do not play inbound Sink RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio.
Page 164: Network Settings Section
ATA Voice Field Reference Line page NAT Keep Alive Destination that should receive NAT keep alive messages. Dest If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server. The default is $PROXY. Voice tab >...
Page 165: Sip Settings Section
ATA Voice Field Reference Line page Voice tab > Line page > SIP Settings section Field Description SIP Transport The TCP choice provides “guaranteed delivery,” which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent.
Page 166 ATA Voice Field Reference Line page SIP GUID This field is not found in the PAP2T. The Global Unique ID is generated for each line for each device. When it is enabled, the ATA device adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset.
Page 167 ATA Voice Field Reference Line page Restrict Source IP If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines.
Page 168: Call Feature Settings Section
ATA Voice Field Reference Line page Reply 182 On Call When this feature is enabled, your ATA device replies with Waiting a SIP 182 response to the caller if it is already in a call and the phone is off-hook. To use this feature, select yes. Otherwise, keep the default, no.
Page 169: Proxy And Registration Section
ATA Voice Field Reference Line page Conference Bridge Select the maximum number of conference call Ports participants. The range is 3 to 10. The default is 3. This field is found on the SPA2102 and PAP2T only. Voice tab > Line page or Phone page> Proxy and Registration section Proxy SIP proxy server for all outbound requests.
Page 170 ATA Voice Field Reference Line page Ans Call Without Expires value in sec in a REGISTER request. The PAP2T will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec Use DNS SRV Whether to use DNS SRV lookup for Proxy and Outbound Proxy.
Page 171: Subscriber Information Section
ATA Voice Field Reference Line page Voice tab > Line page > Subscriber Information section Display Name Display name for caller ID. User ID Extension number to identify this line. Password Password for this line. Use Auth ID To use the authentication ID and password for SIP authentication, select yes.
Page 172: Supplementary Service Subscription Section
ATA Voice Field Reference Line page Voice tab > Line page > Supplementary Service Subscription section The ATA device provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service.
Page 173 ATA Voice Field Reference Line page Accept Last Serv Enable Accept Last Call Service The default is yes. DND Serv Enable Do Not Disturb Service The default is yes. CID_Serv Enable Caller ID Service The default is yes. CWCID Serv Enable Call Waiting Caller ID Service The default is yes.
Page 174: Audio Configuration Section
ATA Voice Field Reference Line page Secure Call Serv Enable Secure Call Service. The default is yes. Referral Serv Referral Services Codes Enable Referral Service. See the parameter for more details. The default is yes. Feature Dial Serv Feature Dial Services Enable Feature Dial Service.
Page 175 ATA Voice Field Reference Line page Second Preferred Second preferred codec for all calls. (The actual codec Codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: Unspecified, G711u, G711a, G726-16, G726-24, G726- 32, G726-40, G729a, or G723.
Page 176 ATA Voice Field Reference Line page Echo Canc Adapt To enable the echo canceller to adapt, select yes. Enable Otherwise, select no. The default is yes. G726-16 Enable To enable the use of the G.726 codec at 16 kbps, select yes.
Page 177 ATA Voice Field Reference Line page FAX Codec To force the ATA to use a symmetric codec during fax Symmetric passthrough, select yes. Otherwise, select no. The default is yes. DTMF Process To use the DTMF process AVT feature, select yes. Otherwise, select no.
Page 178 ATA Voice Field Reference Line page DTMF Tx Strict This parameter is in effect only when "DTMF Tx Mode" is Hold Off Time: set to "strict," and when"DTMF Tx Method" is set to out-of- band; i.e. either AVT or SIP-INFO. The value can be set as low as 40 ms.
Page 179 ATA Voice Field Reference Line page FAX Enable T38 To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. T.38 is supported by the SPA2102 and the SPA8000. The SPA2102 supports a single T.38 connection. The SPA8000 supports one T.38 connection for each of its four modules (Line 1-2, 3-4, 5-6, and 7-8).
Page 180: Gateway Accounts Section (Spa3102)
ATA Voice Field Reference Line page Gateway Accounts section (SPA3102) Gateway1/2/3/4 The first of 4 gateways that can be specified to be used in the <Dial Plan> to facilitate call routing specification (that overrides the given proxy information). This gateway is represented by gw1 in the <Dial Plan>.
Page 181: Dial Plan Section
ATA Voice Field Reference Line page Voice tab > Line page > Dial Plan section The dial plan determines how the entered digits are processed. On the SPA8800, the Line page includes the dial plan fields as described below. However, on the Phone page, the Dial Plans section provides eight spaces where you can enter up to eight dial plans.
Page 182 ATA Voice Field Reference Line page Dial Plan Dial plan script for this line. The default is (*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: •...
Page 183: Fxs Port Polarity Configuration Section
ATA Voice Field Reference Line page Emergency Comma separated list of emergency number patterns. If Number outbound call matches one of the pattern, SPA will disable hook flash event handling. The condition is restored to normal after the phone is on-hook. Blank signifies no emergency number.
Page 184: Pstn-To-Voip Gateway Setup Section (Spa8800)
ATA Voice Field Reference Line page VoIP Caller Default Index of the dial plan in the dial plan pool to be used as the default. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}. The default is 1. 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 SPA8800...
Page 185: Fxo Timer Values (Sec) Section (Spa8800)
ATA Voice Field Reference Line page Voice tab > Line page > FXO Timer Values (sec) section (SPA8800) PSTN Answer Delay in seconds before auto-answering inbound PSTN Delay calls after the PSTN starts ringing. The range is 0-255. The default is 16. PSTN-To-VoIP Call Limit on the duration of a PSTN-To-VoIP Gateway Call.
Page 186: Pstn Disconnect Detection Section (Spa8800)
ATA Voice Field Reference Line page PSTN Hook Flash The length of the hook flash in seconds. During a PSTN-to- VoIP gateway call, the ATA device 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 187 ATA Voice Field Reference Line page Detect VoIP Long If enabled, SPA will disconnect both call legs when the Silence VoIP side has no voice activity for a duration longer than the length specified in the VoIP Long Silence Duration parameter during a gateway call.
Page 188 ATA Voice Field Reference Line page Disconnect Tone This value is the tone script which describes to the SPA the tone to detect as a disconnect tone. The syntax follows a standard Tone Script with some restrictions. Default value is standard US reorder (fast busy) tone, for 4 seconds. Restrictions: •...
Page 189: International Control Section (Spa8800)
ATA Voice Field Reference Line page Voice tab > Line page > International Control section (SPA8800) FXO Port Desired impedance of the FXO Port. Choose from {600, Impedance 900, 370+620, 270+750||150nF, 220+820||120nF, 370 + 620 || 310nf, 320 + 1050 || 230nf, 370 + 820 || 110 nf, 275 + 780 || 115nf, 120 + 820 || 110nf, 350 + 1000 || 210nf, 0 + 900 || 130nf} The default is 600.
Page 190: Call Forward, Speed Dial, Supplementary Services, And Ring Settings (Spa8000 And Spa8800)
ATA Voice Field Reference Line page Tip/Ring Voltage Choices are {3. 1 , 3.2, 3.35, 3.5} (V). Adjust The default is 3.5 V. Ring Indication Choose from {0, 256, 512, 768, 1024, 1280, 1536, 1792} Delay (ms). The default is 512ms. Operational Loop Choices for mA are: {10, 12, 14, 16).
Page 191: Trunk Group Page (Spa8000)
ATA Voice Field Reference Trunk Group page (SPA8000) Trunk Group page (SPA8000) On the SPA8000, you can use the Voice tab > Trunk Group pages (T1 ... T4) to configure the Trunk Groups. This page includes the following sections: • “Line Enable section”...
Page 192: Sip Settings Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page (SPA8000)> SIP Settings section SIP Transport The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent.
Page 193 ATA Voice Field Reference Trunk Group page (SPA8000) SIP GUID This field is not found in the PAP2T. The Global Unique ID is generated for each line for each device. When it is enabled, the ATA device adds a GUID header in the SIP request.
Page 194 ATA Voice Field Reference Trunk Group page (SPA8000) Restrict Source IP If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines.
Page 195: Subscriber Information Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page > (SPA8000) > Subscriber Information section Display Name Display name for caller ID. User ID Extension number for this line. Password Password for this line. Use Auth ID To use the authentication ID and password for SIP authentication, select yes.
Page 196 ATA Voice Field Reference Trunk Group page (SPA8000) Contact List This parameter determines which trunk lines to ring on an incoming call. When an incoming call is detected by the Trunk SUA (SIP User Agent), the SUA first checks if there is capacity to handle the call.
Page 197: Dial Plan Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page (SPA8000) > Dial Plan section Field Description Dial Plan Dial plan script for this trunk. NOTE: The trunk SUA will also apply the Trunk Dial Plan on the number before sending out INVITE to the ITSP.
Page 198: Proxy And Registration Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page (SPA8000) > Proxy and Registration section Proxy SIP proxy server for all outbound requests. Use Outbound Enablse the use of an Outbound Proxy . If set to no, the Outbound Proxy Use OB Proxy in Dialog Proxy...
Page 199: Pstn Line Page (Spa3102)
ATA Voice Field Reference PSTN Line page (SPA3102) Use DNS SRV Whether to use DNS SRV lookup for Proxy and Outbound Proxy. The default is no. DNS SRV Auto If enabled, the PAP2T will automatically prepend the Proxy Prefix or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.
Page 200: Line Enable Section
ATA Voice Field Reference PSTN Line page (SPA3102) • “Network Settings section” section on page 201 • “SIP Settings section” section on page 202 • “Proxy and Registration section” section on page 205 • “Subscriber Information section” section on page 206 •...
Page 201: Network Settings Section
ATA Voice Field Reference PSTN Line page (SPA3102) NAT Keep Alive To send the configured NAT keep alive message Enable periodically, select yes. Otherwise, select no. The default is no. NAT Keep Alive Enter the keep alive message that should be sent periodically to maintain the current NAT mapping.
Page 202: Sip Settings Section
ATA Voice Field Reference PSTN Line page (SPA3102) Network Jitter Determines how jitter buffer size is adjusted by the ATA Level device. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings.
Page 203 ATA Voice Field Reference PSTN Line page (SPA3102) SIP Remote-Party- To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. The default is yes. SIP GUID This field is not available with the PAP2T. The Global Unique ID is generated for each line for each device.
Page 204 ATA Voice Field Reference PSTN Line page (SPA3102) Restrict Source IP If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines.
Page 205: Proxy And Registration Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page (SPA3102) > Proxy and Registration section Proxy SIP proxy server for all outbound requests. Use Outbound Enable the use of Outbound Proxy . If set to no, the Outbound Proxy Use OB Proxy in Dialog Proxy...
Page 206: Subscriber Information Section
ATA Voice Field Reference PSTN Line page (SPA3102) Use DNS SRV Whether to use DNS SRV lookup for Proxy and Outbound Proxy. The default is no. DNS SRV Auto If enabled, the PAP2T will automatically prepend the Proxy Prefix or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.
Page 207: Audio Configuration Section
ATA Voice Field Reference PSTN Line page (SPA3102) Use Auth ID To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. The default is no. Auth ID Authentication ID for SIP authentication. Call Capacity Maximum number of calls allowed on this line interface.
Page 208 ATA Voice Field Reference PSTN Line page (SPA3102) Use Pref Codec To use only the preferred codec for all calls, select yes. Only (The call fails if the far end does not support this codec.) Otherwise, select no. The default is no. Silence Threshold Select the appropriate setting for the threshold: high, medium, or low.
Page 209 ATA Voice Field Reference PSTN Line page (SPA3102) FAX CNG Detect To enable detection of the fax Calling Tone (CNG), select Enable yes. Otherwise, select no. The default is yes. G726-40 Enable To enable the use of the G726 codec at 40 kbps, select yes.
Page 210 ATA Voice Field Reference PSTN Line page (SPA3102) Hook Flash Tx Select the method for signaling hook flash events: None, Method AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO with the single line signal=hf in the message body.
Page 211: Dial Plans Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page (SPA3102) > Dial Plans section Dial Plan 1/2/3/4/5/ Dial plan script for this line. 6/7/8 The default is (xx.) Dial plans in the dial plan pool to be associated with a VoIP Caller or a PSTN Caller.
Page 212 ATA Voice Field Reference PSTN Line page (SPA3102) Line 1 VoIP Caller 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 during normal operation (in other words, not due to fallback to PSTN service when Line 1 VoIP service is down).
Page 213: Voip Users And Passwords (Http Authentication) Section
ATA Voice Field Reference PSTN Line page (SPA3102) VoIP Caller 1/2/3/4/ Index of the dial plan in the dial plan pool to be associated VoIP 5/6/7/8 DP with the VoIP caller who enters the PIN that matches Caller 1/2/3/4/5/6/7/8 PIN The default is 1.
Page 214: Ring Settings Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page (SPA3102) > Ring Settings section Default Ring 1-8, Follow Line Cfg Voice tab > PSTN Line page (SPA3102) > FXO (PSTN) Timer Values (sec) section VoIP Answer Delay Delay in seconds before auto-answering inbound VoIP calls for the FXO account.
Page 215 ATA Voice Field Reference PSTN Line page (SPA3102) PSTN Ring Thru Delay in seconds before starting to ring thru Line 1 after the Delay PSTN starts ringing. In order for Line 1 to have the caller-id information, the delay should be set to larger than the delay required to complete the PSTN caller-id delivery (such as 5s).
Page 216: Pstn Disconnect Detection Section
ATA Voice Field Reference PSTN Line page (SPA3102) PSTN Ring Thru Specify the delay before incoming PSTN calls will ring Line CWT Delay 1 using a Call Waiting Tone. The default is 3. PSTN Ring Timeout Specify the delay after a ring burst before the Gateway decides that the PSTN ring has ended.
Page 217 ATA Voice Field Reference PSTN Line page (SPA3102) Detect Disconnect If enabled, SPA will disconnect both call legs when it Tone detects the disconnect tone from the PSTN side during a gateway call. Disconnect tone is specified in the Disconnect Tone parameter, which depends on the region of the PSTN service.
Page 218 ATA Voice Field Reference PSTN Line page (SPA3102) Disconnect Tone This value is the tone script which describes to the SPA the tone to detect as a disconnect tone. The syntax follows a standard Tone Script with some restrictions. Default value is standard US reorder (fast busy) tone, for 4 seconds.
Page 219: International Control (Settings) Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page (SPA3102) > International Control (Settings) section FXO Port Desired impedance of the FXO Port. Choose from {600, Impedance 900, 370+620, 270+750||150nF, 220+820||120nF, 370 + 620 || 310nf, 320 + 1050 || 230nf, 370 + 820 || 110 nf, 275 + 780 || 115nf, 120 + 820 || 110nf, 350 + 1000 || 210nf, 0 + 900 || 130nf} The default is 600.
Page 220 ATA Voice Field Reference PSTN Line page (SPA3102) Operational Loop Choices for mA are: {10, 12, 14, 16). Current Min The default is 10. On-Hook Speed Choose from {Less than 0.5ms, 3ms (ETSI), 26ms (Australia)}. The default is Less than 0.5ms. Current Limiting Enable or disable current limiting.
Page 221: User Page
ATA Voice Field Reference User page User page Depending on the model of ATA device, there may be one or more User pages. You can use this page to configure the user settings. On the SPA8000, these settings can be configured on the Line pages (Line 1, Line 2, and so on).
Page 222: Call Forward Settings Section
ATA Voice Field Reference User page Voice tab > User page > Call Forward Settings section Cfwd All Dest Forward number for Call Forward All Service In addition to normal call forward destination as used in the other ATAs, on the SPA3102, you can specify the following additional parameters: gw0 –...
Page 223: Selective Call Forward Settings Section
ATA Voice Field Reference User page Voice tab > User page > Selective Call Forward Settings section Cfwd Sel 1 - 8 Caller Caller number pattern to trigger Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8. The default is blank.
Page 224: Supplementary Service Settings Section
ATA Voice Field Reference User page Voice tab > User page > Supplementary Service Settings section The ATA device provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service.
Page 225: Distinctive Ring Settings Section
ATA Voice Field Reference User page Accept Media Controls how to handle incoming requests for loopback Loopback Request operation. Choices are: Never, Automatic, and Manual, where: • never—never accepts loopback calls; reply 486 to the caller • automatic—automatically accepts the call without ringing •...
Page 226: Ring Settings Section
ATA Voice Field Reference User page Voice tab > User page > Ring Settings section Default Ring Default ringing pattern, 1 – 8, for all callers. The default is 1. Default CWT Default CWT pattern, 1 – 8, for all callers. The default is 2.
Page 227: Pstn User Page (Spa3102)
ATA Voice Field Reference PSTN User page (SPA3102) Ring On No New If enabled, the ATA device will play a ring splash when the VM server sends SIP NOTIFY message to the ATA device indicating that there are no more unread voice mails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp.
Page 228: Pstn-To-Voip Speed Dial Settings Section
ATA Voice Field Reference PSTN User page (SPA3102) Voice tab > PSTN User page (SPA3102) > PSTN-To-VoIP Speed Dial Settings section Speed Dial 1-9 The VoIP number to call when the PSTN caller dials a single digit ‘2’ Voice tab > PSTN User page (SPA3102) > PSTN Ring Thru Line 1 Distinctive Ring Settings section Ring1-8 Caller Eight PSTN Caller Number Patterns such that the...
Page 229: Appendix C: Troubleshooting
Troubleshooting This appendix provides solutions to problems that may occur during the installation and operation of the ATA devices. If you can't find an answer here, visit www.cisco.com/en/US/products/ps10024/ NOTE tsd_products_support_series_home.html Q. I want to use a different computer to access the administration web server. The address I entered did not work.
Page 230 Troubleshooting Q. I’m trying to access the ATA administration web server, but I do not see the login screen. Instead, I see a screen saying, “404 Forbidden.” A. If you are using Windows Explorer, perform the following steps until you see the administration web server login screen (Mozilla requires similar steps).
Page 231 Troubleshooting Q. How do I access the ATA device if I forget my password? A. 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.
Page 232 Troubleshooting 4. Add a STUN server to allow traversal of UDP packets through the NAT device. STUN Enable On the SIP tab of the administration web server, set to yes, and STUN Server enter the IP address of the STUN server in 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...
Page 233: Appendix D: Environmental Specifications
Environmental Specifications This appendix provides the specifications for the following ATAs: • “PAP2T,” on page 233 • “SPA2102,” on page 234 • “SPA3102,” on page 234 • “SPA8000,” on page 235 • “SPA8800,” on page 235 • “WRTP54G,” on page 236 PAP2T Device 3.98”...
Page 234: Spa2102
Environmental Specifications SPA2102 SPA2102 Device 3.98” x 3.98” x 1. 1 0” (101 x 101 x 28 mm) W x H x D Dimensions Unit Weight 5.29 oz (0. 1 5kg) Power 100-240V 50-60Hz (26-34VA), AC Input Certification FCC (Part 15 Class B), CE, ICES-003 Operating 32º...
Page 235: Spa8000
Environmental Specifications SPA8000 SPA8000 Device 6.69” x 1.54” x 8.66” (170 x 39 x 220 mm) Dimensions Unit Weight 2.85 lbs (1.30kg) Power 100-240V 50-60Hz (26-34VA), AC Input Certification FCC (Part 15 Class B), CE, ICES-003, A-Tick Certification, RoH, UL Operating 32º...
Page 236: Wrtp54G
Environmental Specifications WRTP54G WRTP54G Device 6.69 ” x 6.69” x 1.22” (170 x 170 x 31 mm) Dimensions Unit Weight 13.60 oz (.39 kg) Power External, 12V DC, 1.0A Certification FCC (Part 15 Class B), CE, UL Operating 32º to 104º F(0 to 40ºC) Temp Storage -4º...
Page 237: Appendix E: Where To Go From Here
Where to Go From Here This appendix describes additional resources that are available to help you and your customer obtain the full benefits of the SPA9000 Voice System. • “Product Resources,” on page 237 • “Related Documentation,” on page 238 Product Resources Website addresses in this document are listed without http:// in front of the address because most current web browsers do not require it.
Page 238: Related Documentation
Where to Go From Here Related Documentation Resource Location www.cisco.com/go/osln Open Source License Notices http://www.cisco.com/en/US/products/ps10024/ Regulatory Compliance and prod_maintenance_guides_list.html Safety Information Cisco Partner www.cisco.com/web/partners/sell/smb Central (Login Required) Cisco Small www.cisco.com/smb Business Home Related Documentation The following table describes the various documents that Cisco provides to help you to install, configure, and manage the SPA9000 Voice System and its components.
Page 239 Where to Go From Here Related Documentation Document Title Description Intended Audience • Administration and SPA9000 Voice System VARs and Service configuration of system Administration Guide Providers features using the SPA9000 and SPA400 • Deployment options for ITSP, PSTN, and ISDN services •...
Page 240 ATA Variables--DELETE this file FROM PDF before publishing During Drafting (when you want to see condition indicators): On the menu, choose Special > Variables. Click Title, and then click Edit. Verify STEP 1 that it shows the correct title of this book. On the menu, choose Special >...
Page 241 d. Click Deselect All to clear all of the check boxes, and then check only the following check boxes: Variable Defintions Conditional Text Settings e. Click Import. Generate/update your book to ensure that all page numbers, chapter numbers, STEP 3 etc., are updated.

This manual is also suitable for:

Spa2102 Spa3102 Spa8000 Spa8800 Pap2t

Cisco SPA2102-AN - Single Port Router Administration Manual

Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters

Chapter 2: Basic Administration and Configuration

Chapter 3: Configuring Your System for ITSP Interoperability

Chapter 4: Configuring Voice Services

Chapter 5: Configuring Music on Hold

Chapter 6: Configuring the PSTN (FXO) Gateway on the SPA3102

Appendix A: ATA Routing Field Reference

Appendix B: ATA Voice Field Reference

Appendix C: Troubleshooting

Appendix D: Environmental Specifications

Appendix E: Where to Go from here

Quick Links

Troubleshooting

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Summary of Contents for Cisco SPA2102-AN - Single Port Router