Page 1 ADMINISTRATION GUIDE Cisco Small Business Pro SPA2102, SPA3102, SPA8000, PAP2T, WRP400 Analog Telephone Adapters...
Page 2 OL-17901-01...
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 ATA Software Features Voice Supported Codecs SIP Proxy Redundancy Other ATA Software Features...
Page 4 Contents Reboot URL Provisioning Your ATA Device 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 Whether the Router Uses Symmetric or Asymmetric NAT Firewalls and SIP...
Page 5 Contents Using a Mini-Certificate Generating a Mini Certificate SIP Trunking and Hunt Groups on the SPA8000 About SIP Trunking Setting the Trunk Group Call Capacity Inbound Call Routing for a Trunk Group Contact List for a Trunk Group Outgoing Call Routing for a Trunk Group Configuring a Trunk Group Trunk Group Management Setting the Hunt Policy...
Page 6 Contents 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 Call Forwarding to PSTN Gateway Appendix A: ATA Routing Field Reference Router Status page...
Page 7 Contents Port Forwarding Settings section DMZ Settings section Miscellaneous Settings section System Reserved Ports Range section Appendix B: ATA Voice Field Reference Info page Product Information section System Status section Line Status section System Information section (PAP2T) PSTN Line Status section (SPA3102) Trunk Status section (SPA8000) System page System Configuration section...
Page 8 Contents Distinctive Ring/CWT Pattern Names section Ring and Call Waiting Tone Spec section Control Timer Values (sec) section Vertical Service Activation Codes section Vertical Service Announcement Codes section (SPA2102, SPA8000) Outbound Call Codec Selection Codes section Miscellaneous section Line page Line Enable section Streaming Audio Server (SAS) section NAT Settings section...
Page 9 Contents PSTN Line page (SPA3102) Line Enable section NAT Settings section Network Settings section SIP Settings section Proxy and Registration section Subscriber Information section 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...
Page 10 Contents Appendix C: Provisioning Reference (WRP400) Appendix D: Troubleshooting Appendix E: Environmental Specifications PAP2T SPA2102 SPA3102 SPA8000 WRP400 WRTP54G Appendix F: Where to Go From Here Product Resources Related Documentation Appendix G: Additional Information Appendix H: Support Contacts ATA Administration Guide viii...
Page 11: About This Document
“Finding Information in PDF Files,” on page xiii Purpose This document provides information that administrators can use to configure and manage Cisco ATAs that are used in conjunction with the SPA9000 Voice System. Audience This document is written for the following audience: •...
Page 12 Preface Firmware This guide describes the features that are available in the following firmware releases. Product Firmware Version PAP2T 5. 1 .6 SPA2102 5.2.5 SPA3102 5. 1 .7 SPA8000 6. 1 .3 WRP400 1.00.06 Document Conventions The following are the typographic conventions used in this document. Typographic Meaning Element...
Page 13 The information in this guide is organized into the following chapters and appendices: Chapter Contents Chapter 1, “Introducing This chapter introduces the functionality of the ATA Cisco Small Business devices and describes the features that are Analog Telephone available. Adapters” Chapter 2, “Basic...
Page 14 Preface Chapter Contents Appendix D, This appendix provides solutions to problems that “Troubleshooting” may occur during the installation and operation of the ATA devices. Appendix F, “Where to Go These appendices provide information about other From Here” resources that may be useful to you. Appendix G, “Additional Information”...
Page 15 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 16 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 17 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 18: Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters
Introducing Cisco Small Business Analog Telephone Adapters This guide describes the administration and use of Cisco Small Business analog telephone adapters (ATAs). These ATA devices are a key element in the end-to- end IP Telephony solution. An ATA device provides user access to Internet phone services through one or more standard telephone RJ-11 phone ports using standard analog telephone equipment.
Page 19: Comparison Of Ata Devices
Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Comparison of ATA Devices Each ATA device is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP Telephony services delivered over broadband or high-speed Internet connections. An ATA device maintains the state of each call it terminates and makes the proper reaction to user input events (such as on/off hook or hook flash).
Page 20 Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices The information contained in this guide is not a warranty from Cisco. Customers NOTE planning to use ATA devices in a VoIP service deployment are advised to test all functionality they plan to support before putting the ATA device in service.
Page 21 Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Figure 1 How ATAs Provide Voice Connectivity Ethernet/Wireless WRP400, WRTP54G, Fax (up to 4 and SPA2102 SPA8000) DSL/cable modem Broadband router Internet SPA8000, Broadband PAP2T router Analog phone (up to 8 with...
Page 22: 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 23: 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...
Page 24: 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...
Page 25: 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...
Page 26: 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...
Page 27: Ata Software Features
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features NOTE • With the SPA8000, use line 1 or line 2 to access the IVR functions. See the SPA8000 Quick Installation Guide for IVR instructions. • For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the SPA8000 is located behind a router.
Page 28 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features codec is to be used for each line. G.711a and G.711u are always enabled. Configure your preferred codec in the (FXS) tab in the Administration Web Server. “ATA Voice Field Reference,” on page121.
Page 29: 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 30 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 31 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description Adjustable Audio This feature allows the user to set the number of audio Frames Per Packet frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality.
Page 32 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 33 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 34 Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features Feature Description SIP Over TLS SPA2102, SPA3102, and WRP400 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 35 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 36 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 37: 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 35 •...
Page 38: Downloading Firmware
Downloading Firmware Always download and install the latest firmware for your ATA device before doing any configurations. You can find the latest firmware at www.cisco.com/go/ smallbiz. Basic Installation and Configuration See your the Quick Installation Guide and the User Guide the ATA model that you are installing.
Page 39: Setting Up Your Ata Device
Basic Administration and Configuration Setting up Your ATA Device Use the administration computer to install the latest firmware: STEP 3 a. Extract the Zip file, and then run the executable file to upgrade the firmware. Firmware Upgrade Warning b. When the window appears, click Continue.
Page 40: Using The Administration Web Server
Basic Administration and Configuration Using the Administration Web Server 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. The parameters that the User account can access are specified using the Administrator account on the Provisioning page of the administration web server.
Page 41: Connecting To The Administration Web Server
Basic Administration and Configuration Using the Administration Web Server 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 42 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 43: 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 44: Advanced Configurations
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device If the device has more than one Line tab, each line tab must be configured NOTE separately. Each line tab can be configured for a different ITSP. Advanced Configurations Other parameters may need to be changed from the defaults, depending on the requirements of a specific ITSP.
Page 45: Resync Url
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device If the value of the Upgrade Enable parameter in the Provisioning page is No, you NOTE cannot upgrade the ATA device even if the web page indicates otherwise. The syntax of the Upgrade URL is as follows: http://spa-ip-addr/admin/upgrade?[protocol://][server-name[:port]][/ firmware-pathname] Both HTTP and TFTP are supported for the upgrade operation.
Page 46: Reboot Url
URL into the device. The ATA device accepts profiles in XML format, or alternatively in a proprietary binary format, which is generated by a profile compiler tool available from Cisco. Find the Profiler Compiler for your ATA at http://www.cisco.com/web/partners/ sell/smb/products/voice_and_conferencing.html#~vc_technical_resources.
Page 47: Configuration Profile
(HTTPS), or it can resync to a binary profile generated by the Cisco-supplied profile compiler. In the latter case, the profile compiler can encrypt the profile specifically for the target ATA device, without requiring an explicit key exchange.
Page 48 Basic Administration and Configuration Provisioning Your ATA Device The names of parameters in XML profiles can generally be inferred from the ATA configuration Web pages, by substituting underscores (_) for spaces and other control characters. Further, to distinguish between Lines 1, 2, 3, and 4, corresponding parameter names are augmented by the strings _1_, _2_, _3_, and _4_.
Page 49: 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 50: 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 51 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 52: 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 53 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 54: Determining Whether The Router Uses Symmetric Or Asymmetric Nat
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Determining Whether the Router Uses Symmetric or Asymmetric NAT 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 55: 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 56: 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 (SPA2102, SPA3102 or SPA8000),” on page 55 • “Managing Caller ID Service,” on page 58 •...
Page 57: Using A Fax Machine (Spa2102, Spa3102 Or Spa8000)
Configuring Voice Services Using a FAX Machine (SPA2102, SPA3102 or SPA8000) • G.726-40 • G.729a • G.723 WRTP54G • G.711u (configured by default) • G.711a • G.726-32 • G.729a • G.723 WRP400 • G.711u (configured by default) • G.711a • G.726-32 •...
Page 58 • 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. For example: modem passthrough nse payload-type 110 codec g711ulaw...
Page 59: Fax Troubleshooting
STEP 8 • If you are an end user of VoIP products, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment. • If you are an authorized Cisco partner, contact Cisco technical support. ATA Administration Guide...
Page 60: 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 61 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 62: 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 63: 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 64 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 65 Configuring Voice Services Configuring Dial Plans Digit Sequence Function Enter a comma between digits to play an “outside (comma) 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 66 Configuring Voice Services Configuring Dial Plans • Local dialing with seven-digit number EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111) After a user presses 9, an external dial tone sounds.
Page 67 Configuring Voice Services Configuring Dial Plans • Blocked number EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! 9, 011xxxxxx. | 0 | [49]11 ) This digit sequence is useful if you want to prevent users from 9, 1 900 xxxxxxx ! dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S..
Page 68 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 69 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 70 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 71 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 72: 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 73 Configuring Voice Services Configuring Dial Plans Enter the desired values in the Interdigit Long Timer field and the Interdigit Short STEP 4 Timer field. Refer to the definitions at the beginning of this section. ATA Administration Guide...
Page 74: 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 72 • ”Secure Call Details” section on page 73 •...
Page 75: Secure Call Details
Configuring Voice Services Secure Call Implementation The signing agent is implicit and must be the same for all ATAs that communicate securely with each other. The public key of the signing agent is pre-configured into the ATA device by the administrator and is used by the ATA device to verify the Mini-Certificate of its peer.
Page 76: Using A Mini-Certificate
Configuring Voice Services Secure Call Implementation The caller sends the “Caller Final” message to the called party with the following STEP 2 information: • Message ID (4B) • Encrypted Master Key (16B or 128b) • Encrypted Master Salt (16B or 128b) Using a Mini-Certificate The Master Key and Master Salt are encrypted with the public key from the called party mini-certificate.
Page 77: Generating A Mini Certificate
Secure Call Implementation Generating a Mini Certificate Cisco provides a Mini Certificate Generator for the generation of mini certificates and private keys. Partners can download the Mini Certificate Generator by going to Cisco Partner Central, Voice & Conferencing page, Technical Resources section.
Page 78 Configuring Voice Services Secure Call Implementation EXAMPLE: gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34” This example produces the following Mini Certificate and SRTP Private Key: <Mini Certificate> Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEw MTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/ xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhx ES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/ LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE 9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBx uMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs= <SRTP Private Key> b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/ e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ== ATA Administration Guide...
Page 79: 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 80: 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 81 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 82: 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 83: 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 84 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 85: Outgoing Call Routing For A Trunk Group
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 • ?,hunt=ra;12;1,cfwd=14085550123 A wildcard character is used to represent “all trunk lines.” The Trunk UA 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 86: 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 87: 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 88: Setting The Hunt Policy
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 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 89: Additional Notes About Trunk Groups
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Additional Notes About Trunk Groups This section includes information about other topics that may be of interest when you are configuring trunk groups: • Voice mail: There is no individual mail box for a trunk line. For example, if lines 1, 2, 3, and 4 belong the trunk group T1, then the four lines implicitly share the same voice mail box from the ITSP.
Page 90: 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 88 •...
Page 91: 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 92: 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 93 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 94: 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 95: 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 96: 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. • ”Connecting to PSTN and VoIP Services” section on page 94 • ”How VoIP-To-PSTN Calls Work” section on page 95 •...
Page 97: How Voip-To-Pstn Calls Work
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work • VoIP user—VoIP caller that has a user account (user-id and password) on the ATA device • PSTN caller—One who calls the ATA device from the PSTN to obtain VoIP service Line 1 can be configured with a regular VoIP account and can be used in the same way as the Line 1 of any ATA device.
Page 98 Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work If the FXO port is currently in use (off-hook) or the PSTN line is being used by another extension, the ATA device replies to the INVITE with a 503 response. Dialed-Number User ID Otherwise, it compares the <...
Page 99: Two-Stage Dialing
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work If Authentication is disabled, a default dial plan is used for all unknown VoIP users. NOTE Two-Stage Dialing In two-stage dialing, the ATA device takes the FXO port off-hook but does not automatically dial any digits after accepting the call.
Page 100: How Pstn-To-Voip Calls Work
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work Table 2 Parameters for Two-Stage Dialing Parameter Description Values Page VoIP Caller 1/2/ PSTN The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, 31-character string 3/4/5/6/7/8 PIN Line or 8.
Page 101: Terminating Gateway Calls
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work Terminating Gateway Calls There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call is terminated when either call leg is ended. When the call terminates, the FXO port goes on-hook so the PSTN line can be used again.
Page 102 Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work Parameter Description Values Page Disconnect PSTN Tone Script of the disconnect tone to ToneScript Tone: Line detect. The ATA device supports two The default frequency components. If the tone has is 480@- only one frequency, use the same value 30,620@-...
Page 103: Voip Outbound Call Routing
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work VoIP Outbound Call Routing Calls made from Line 1 are routed through the configured Line 1 service provider, by default. You can override this behavior by IP dialing, through which the calls can be routed to any IP address entered by the user.
Page 104: Configuring Voip Failover To Pstn
Dial 8 to start outside dial tone, prepend 1408 <8,:1408>xxxxxxx<:@pstn. cisco.com:5061;usr=joe; followed by seven digits, and route the call to pwd=joe_pwd;nat> pstn.cisco.com:5061, with user-id = joe, and pwd = bell_pwd, and enable NAT mapping Dial 8 to start outside dial tone, prepend 1408 <8,:1408>xxxxxxx<:@gw2:5061; usr=”Alex Bell”;pwd= followed by seven digits, and route the call to ”anything”;nat=no>...
Page 105: Sharing One Voip Account Between The Fxs And Pstn Lines
Configuring the PSTN (FXO) Gateway on the SPA3102 Sharing One VoIP Account Between the FXS and PSTN Lines Parameter Description Value Page Auto PSTN Line 1 If enabled, the ATA device automatically Fallback routes outbound calls to Gateway 0 when default is registration fails or network link is down.
Page 106: Other Options
Configuring the PSTN (FXO) Gateway on the SPA3102 Other Options Other Options This section describes other options provided by the ATA device. It includes the following topics: • ”PSTN Call to Ring Line 1” section on page104 • ”Symmetric RTP” section on page104 •...
Page 107: Call Progress Tones
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios Call Progress Tones The ATA has configurable call progress tones. Call progress tones are generated locally on the ATA, so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance a dial tone played back to an end user) may include: •...
Page 108: 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 109 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 110 Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios Without Authentication This scenario can also be implemented without authentication, using one-stage or two-stage dialing, as in the HTTP Authentication case. The default VoIP caller dial plan is used in this scenario. Authentication is performed when the method is none VoIP or when the source IP address of the inbound INVITE matches one of the Access List...
Page 111: Call Forwarding To Pstn Gateway
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios Call Forwarding to PSTN Gateway This section describes a number of scenarios that forward calls to the PSTN gateway. It includes the following topics: • ”Forward-On-No-Answer to the PSTN Gateway” section on page109 •...
Page 112 Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios Forward to a Particular PSTN Number target-number>@gw0 In this scenario, the forward destination is set to < >. This is the same as in the previous examples, except that the ATA device automatically dials the given target number on the PSTN line right after it answers the VoIP call leg.
Page 113: 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, and SPA8000 routers. To NOTE configure router settings for the PAP2T, WRP400, and WRTP54G, see the user guide for the router.
Page 114: 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 115: Wan Setup Page
ATA Routing Field Reference WAN Setup 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 116: Static Ip Settings Section
ATA Routing Field Reference WAN Setup 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 117: Optional Settings Section
ATA Routing Field Reference WAN Setup 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 118: Mac Clone Settings Section
ATA Routing Field Reference WAN Setup 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 119: Vlan Settings Section
ATA Routing Field Reference LAN Setup 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 120: 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 121: Port Forwarding Settings Section
ATA Routing Field Reference Application page 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 Enable forwarding for the chosen application. Options are Yes or No.
Page 122: 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 123: 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 124: 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 page122 •...
Page 125: 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 126 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 127 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 128: 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 129 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 130 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 131: 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 132: 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 page130 •...
Page 133: 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 134: 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 135: 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 136 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 137: 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 138 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 139: 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 140: 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 141 Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/ATA device-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the ATA device’s local time, not the time reported by an NTP server. If the ATA device receives a RR from the peer, it attempts to compute the round trip delay and show it as the <Call Round Trip Delay>...
Page 142: 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 143: 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 144 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 145 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 146: 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 147: 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 148: 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 played automatically when Dial Tone or any of its alternatives times out. The default is 350@-19,440@-19;10(*/0/1+2).
Page 149 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 150: 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 151: 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 152: 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 153: 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 154 ATA Voice Field Reference Regional page Hook Flash Timer Maximum on-hook time before off-hook qualifies as hook- flash. More than this the on-hook event is treated as on- hook (no hook-flash event). Range: 0.4–1.6 seconds. The default is 0.9. Callee On Hook Phone must be on-hook for at this time in sec before the Delay ATA device will tear down the current inbound call.
Page 155: Vertical Service Activation Codes Section
ATA Voice Field Reference Regional page CPC Delay Delay in seconds after caller hangs up when the ATA device starts removing the tip-and-ring voltage to the attached equipment of the called party. Range: 0–255 seconds. ATA device has had polarity reversal feature since release 1.0 which can be applied to both the caller and the callee end.
Page 156 ATA Voice Field Reference Regional page Blind Transfer Code Begins a blind transfer of the current call to the extension specified after the activation code. The default is *98. Call Back Act Code Starts a callback when the last outbound call is not busy. The default is *66.
Page 157 ATA Voice Field Reference Regional page Block Last Act Blocks the last inbound call. Code The default is *60. Block Last Deact Cancels blocking of the last inbound call. Code The default is *80. Accept Last Act Accepts the last outbound call. It lets the call ring through Code when do not disturb or call forwarding of all calls are enabled.
Page 158 ATA Voice Field Reference Regional page DND Act Code Enables the do not disturb feature. The default is *78. DND Deact Code Disables the do not disturb feature. The default is *79. CID Act Code Enables caller ID generation. The default is *65. CID Deact Code Disables caller ID generation.
Page 159 ATA Voice Field Reference Regional page Attn-Xfer Act Code If the code is specified, the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer. Modem Line Toggle Toggles the line to a modem. Code The default is *99.
Page 160 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 161: 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 162 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 163: 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 164 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 165 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 166 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 167: 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 168: Line Enable Section
ATA Voice Field Reference Line page The SPA2102 provides one User tab for each Line tab (User 1 and User 2), where many of the line-specific configuration parameters are contained. The SPA8000 does not provide User tabs, but consolidates all the line-specific parameters on the Line tab.
Page 169: 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 170: 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 171: 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 172 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 173 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 174: Call Feature Settings Section
ATA Voice Field Reference Line page Use Anonymous When set to yes, use “anonymous” in the SIP message with RPID when remote party ID is requested in the SIP message. This field is found on the SPA2102 only. Default is yes. Use Local Addr in The IP address of the local address enclosed in the FROM FROM...
Page 175: Proxy And Registration Section
ATA Voice Field Reference Line page Voice tab > Line page > Proxy and Registration section Proxy SIP proxy server for all outbound requests. Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Use Outbound Outbound Proxy Enablse the use of an...
Page 176: Subscriber Information Section
ATA Voice Field Reference Line page 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 177: Supplementary Service Subscription Section
ATA Voice Field Reference Line page 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. Directory Number Enter the number for this line.
Page 178 ATA Voice Field Reference Line page Block CID Serv Enable Block Caller ID Service. The default is yes. Block ANC Serv Enable Block Anonymous Calls Service The default is yes. Dist Ring Serv Enable Distinctive Ringing Service The default is yes. Cfwd All Serv Enable Call Forward All Service The default is yes.
Page 179 ATA Voice Field Reference Line page Call Back Serv Enable Call Back Service. Three Way Call Enable Three Way Calling Service. Three Way Calling is Serv required for Three Way Conference and Attended Transfer. The default is yes. Three Way Conf Enable Three Way Conference Service.
Page 180: Audio Configuration Section
ATA Voice Field Reference Line page Voice tab > Line page > Audio Configuration section 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.
Page 181: Voip Fallback To Pstn Section (Spa3102)
ATA Voice Field Reference Line page Voice tab > Line page > VoIP Fallback to PSTN section (SPA3102) Auto PSTN Fallback If enabled, the ATA device automatically routes all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down).
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 Trunk Group page (SPA8000) 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: Line Enable Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page > Line Enable section Line Enable To enable this line for service, select yes. Otherwise, select The default is yes. Voice tab > Trunk Group page > Network Settings section SIP ToS/DiffServ TOS/DiffServ field value in UDP IP packets carrying a SIP...
Page 185 ATA Voice Field Reference Trunk Group page (SPA8000) SIP Port Port number of the SIP message listening and transmission port. The default is 5060. SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes.
Page 186 ATA Voice Field Reference Trunk Group page (SPA8000) 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 187 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 188: Subscriber Information Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page 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 189 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 190: Dial Plan Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page > 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 191: Proxy And Registration Section
ATA Voice Field Reference Trunk Group page (SPA8000) Voice tab > Trunk Group page > Proxy and Registration section Proxy SIP proxy server for all outbound requests. Use Outbound Outbound Proxy Enablse the use of an . If set to no, the Outbound Proxy Use OB Proxy in Dialog Proxy...
Page 192: 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 193: Line Enable Section
ATA Voice Field Reference PSTN Line page (SPA3102) • ”Network Settings section” section on page192 • ”SIP Settings section” section on page193 • ”Proxy and Registration section” section on page195 • ”Subscriber Information section” section on page197 • ”Audio Configuration section” section on page198 •...
Page 194: 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 195: 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 196 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 197: Proxy And Registration Section
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 198 ATA Voice Field Reference PSTN Line page (SPA3102) Outbound Proxy Use Outbound Enable the use of . If set to no, the Outbound Proxy Use OB Proxy in Dialog Proxy parameter and ignored. The default is no. Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
Page 199: Subscriber Information Section
ATA Voice Field Reference PSTN Line page (SPA3102) 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 200: Audio Configuration Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > Audio Configuration section 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.
Page 201 ATA Voice Field Reference PSTN Line page (SPA3102) G723 Enable To enable the use of the G723a codec at 6.3 kbps, select yes. Otherwise, select no. The default is yes. Echo Canc Adapt To enable the echo canceller to adapt, select yes. Enable Otherwise, select no.
Page 202 ATA Voice Field Reference PSTN Line page (SPA3102) FAX Codec To force the ATA device to use a symmetric codec during Symmetric fax passthrough, select yes. Otherwise, select no. The default is yes. DTMF Process AVT This field is not available for the PAP2T. To use the DTMF process AVT feature, select yes.
Page 203: Dial Plans Section
ATA Voice Field Reference PSTN Line page (SPA3102) FAX Enable T38 To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. The default is yes. FAX Tone Detect This parameter has three possible values: Mode caller or callee - SPA will detect FAX tone whether it is callee or caller...
Page 204: Voip-To-Pstn Gateway Setup Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > VoIP-To-PSTN Gateway Setup section VoIP-To-PSTN Enable or disable VoIP-To-PSTN Gateway functionality. Gateway Enable The default is yes. VoIP Caller Method to be used to authenticate a VoIP Caller to access Authentication the PSTN gateway.
Page 205 ATA Voice Field Reference PSTN Line page (SPA3102) VoIP Caller ID A comma-separated list of caller number templates such Pattern that callers with numbers not matching any of these templates are rejected for PSTN gateway service, regardless of the setting of the authentication method. The comparison is applied before the access list is applied.
Page 206: Voip Users And Passwords (Http Authentication) Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > VoIP Users and Passwords (HTTP Authentication) section VoIP User 1/2/3/4/ The first of 8 user-id’s that a VoIP Caller can use to 5/6/7/8 Auth ID authenticate itself to the SPA using the HTTP Digest method (in other words, by embedding an Authorization header in the SIP INVITE message sent to the SPA.
Page 207: Ring Settings Section
ATA Voice Field Reference PSTN Line page (SPA3102) VoIP User 1/2/3/4/ Index of the dial plan in the dial plan pool to be used with 5/6/7/8 DP VoIP User 1. The default is 1. VoIP User 1/2/3/4/ The password to be used with VoIP User 1. The user 5/6/7/8 Password assumes the identity of VoIP User 1 must therefore compute the credentials using this password, or the INVITE...
Page 208 ATA Voice Field Reference PSTN Line page (SPA3102) VoIP DLG Refresh Interval between (SIP) Dialog refresh messages sent by Intvl the SPA to detect if the VoIP call-leg is still up. If value is set to 0, SPA will not send refresh messages and VoIP call-leg status is not checked by the SPA.
Page 209: Pstn Disconnect Detection Section
ATA Voice Field Reference PSTN Line page (SPA3102) 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 210 ATA Voice Field Reference PSTN Line page (SPA3102) Detect (PSTN) Long If enabled, SPA will disconnect both call legs when the Silence PSTN side has no voice activity for a duration longer than Long Silence Duration the length specified in the parameter during a gateway call The default is yes.
Page 211 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 212 ATA Voice Field Reference PSTN Line page (SPA3102) Disconnect Tone Spain—425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1) — continued Impedance: 220+820||120nF Portugal—425@-10; 10(0.5/0.5/1)— Impedance:220+820||120nF Poland—425@-10; 10(0.5/0.5/1)— Impedance: n/a Denmark—425@-10; 10(0.25/0.25/1)— Impedance: 600 ATA Administration Guide...
Page 213: International Control (Settings) Section
ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > 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 214 ATA Voice Field Reference PSTN Line page (SPA3102) SPA To PSTN Gain dB of digital gain (or attenuation if negative) to be applied to the signal sent from the SPA to the PSTN side. The range is -15 to 12. The default is 0.
Page 215: User Page
ATA Voice Field Reference User page Ring Threshold Choose from {13.5–16.5, 19.35–2.65, 40.5–49.5} (Vrms). The default is 13.5-16.5 Vrms. Ringer Impedance Choose from {High, Synthesized(Poland, S.Africa, Slovenia)}. The default is high. Line-In-Use Voltage Determines the voltage threshold at which the SPA-3000 assumes the PSTN is in use by another handset sharing the same line (and will declare PSTN gateway service not available to incoming VoIP callers).
Page 216: 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 217: 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 218: 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 219: 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 220: 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 221: Pstn User Page (Spa3102 Only)
ATA Voice Field Reference PSTN User page (SPA3102 Only) 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 222: Pstn Ring Thru Line 1 Distinctive Ring Settings Section
ATA Voice Field Reference PSTN User page (SPA3102 Only) Voice tab > PSTN User page > PSTN Ring Thru Line 1 Distinctive Ring Settings section Ring1-8 Caller Eight PSTN Caller Number Patterns such that the corresponding ring will be used to ring through Line 1 if the PSTN caller matches this pattern.
Page 223 Provisioning Reference (WRP400) This chapter provides information about the parameters that can be provisioned from an XML profile by using the profile compiler tool (SPC). For instructions about provisioning, see the SPA Provisioning Guide in Cisco NOTE Partner Central, http://www.cisco.com/web/partners/sell/smb.
Page 224 Provisioning Reference (WRP400) Feature/XML Parameters Examples RTSP <RTSP>rtsp_enable</RTSP> To enable RTSP: <RTSP>rtsp_enable=1 </ RTSP> <RTSP> rtsp_enable: Real Time Streaming Protocol (RTSP); 1 (enabled) or 0 To disable RTSP: <RTSP>rtsp_enable=0 </ (disabled) RTSP> IGMP <IGMP>force_igmp_version,multicast To specify IGMP version 1 with multicast _pass,multicast_immediate_leave </ pass through and immediate leave: <IGMP>...
Page 225 Provisioning Reference (WRP400) Feature/XML Parameters Examples QoS Category <QOS_PRIORITY_RULE>category_ To configure a rule for an application: Priority Rule number,name, priority,port_range</ <QOS_PRIORITY_RULE>category_num= QOS_PRIORITY_RULE> 1,name= ap1, priority=3,port_range= <QOS_PRIORITY 111;222; 0;333;444;1</QOS_PRIORITY_ _ RULE> category_num: QoS Category RULE> number; 1 (application), 2 (online game), 3 To configure a rule for an online game: (MAC address), 4 (Ethernet port) Format 1 (default game): <QOS_...
Page 226 Provisioning Reference (WRP400) Feature/XML Parameters Examples Basic Wireless <WL_BASIC_SET_1>wl_net_mode,wl To enable SSID-1 and specify the SSID Settings for _closed,wl_ssid</ name: <WL_BASIC_SET_1> wl_net_mode Primary Network WL_BASIC_SET_1> =g-only,wl_closed=0, wl_ssid=aaabbb</ WL_BASIC_SET_1> <WL_BASIC_SET wl_net_mode: Network mode; mixed, _1> b-only, g-only, or disabled To configure SSID-1 as a Wireless B network: <WL_BASIC_SET_1>wl_net_ wl_closed: SSID broadcast status;...
Page 227 Provisioning Reference (WRP400) Feature/XML Parameters Examples Wireless Security <WL_SECURITY_SET_1>wl_security To disable Wireless Security 1: <WL_ for SSID1 _mode2= [mode],[parameters]</ SECURITY_SET_1>wl_security_mode2= <WL_SECURITY_ WL_SECURITY_SET_1> disabled </WL_SECURITY_ SET_1> SET_1> <WL_SECURITY_SET_2>wl 1 _securit To disable Wireless Security 2: <WL_ Wireless Security y_mode2= [mode],[parameters]</ SECURITY_SET_1>wl 1 _security_mode2=di for SSID2 WL_SECURITY_SET_1>...
Page 228 Provisioning Reference (WRP400) Feature/XML Parameters Examples WPA Personal and WPA2 Personal To enable Wireless WPA Personal, specify Parameters the keys and set the renewal rate: <WL_ SECURITY_SET_1>wl_ security_ mode2= wl_crypto: WPA algorithms; tkip wpa_personal,wl_ crypto=aes, wl_wpa_ (TKIP) or aes (AES) psk=personal, wl_wpa_gtk_ rekey=700</ WL_ SECURITY_SET_1>...
Page 229 Provisioning Reference (WRP400) Feature/XML Parameters Examples LAN DHCP <LAN_DHCP>dhcp_lease,dhcp_defa To set the client lease time: <LAN_DHCP> ult_lease</LAN_DHCP> dhcp_default_lease=888 </LAN_DHCP> <LAN_DHCP> dhcp_lease: Client lease time in To set lease time and default lease time: minutes; numerals from 1 to 9999 <LAN_DHCP>dhcp_lease=777,dhcp_ default_lease=888</LAN_DHCP>...
Page 230 Provisioning Reference (WRP400) Feature/XML Parameters Examples WAN Type <WAN_TYPE>wan_proto=[mode], [parameters]</WAN_TYPE> <WAN_TYPE> wan_proto: Internet connection type; dhcp, static, pppoe, pptp, l2tp, heartbeat DHCP Parameters To configure a DHCP connection: <WAN_ TYPE>wan_proto=dhcp No other settings are required. </WAN_TYPE> Static IP Parameters To configure a Static IP connection: <WAN_TYPE>wan_proto=static,wan_ wan_ipaddr: WAN IP address ipaddr=192.
Page 231 Provisioning Reference (WRP400) Feature/XML Parameters Examples Heartbeat for Telstra Cable Network To configure a Telstra Cable connection: Parameters <WAN_TYPE>wan_proto= heartbeat,ppp_username=adc,ppp_ hb_server_ip: Heartbeat server IP passwd=def,hb_ server_ip= 192. 1 68. 0. 1 6</ address WAN_TYPE> ppp_username: User name; enter from 1 to 63 ASCII characters ppp_passwd: Password;...
Page 232 Provisioning Reference (WRP400) Feature/XML Parameters Examples Fail Pattern: <PPP_DEMAND>ppp_demand=1,ppp_ idletime= 66666</PPP_DEMAND> <PPP_DEMAND>ppp_demand=0,ppp_ redialperiod=777</PPP_DEMAND> <PPP_DEMAND>ppp_demand=1 </PPP_ DEMAND> <PPP_DEMAND>ppp_demand=0 </PPP_ DEMAND> <PPP_DEMAND>ppp_demand=1,ppp_ redialperiod=77</PPP_DEMAND> <PPP_DEMAND>ppp_demand=0,ppp_ idletime= 666</PPP_DEMAND> WAN Host <WAN_HOST>wan_hostname=host_test, To specify a WAN hostname and WAN domain wan_domain=domain</WAN_HOST> name: <WAN_HOST> wan_ <WAN_HOST> hostname=host_test,wan_domain= wan_hostname: WAN hostname;...
Page 233 Provisioning Reference (WRP400) Feature/XML Parameters Examples Fail Pattern <WAN_MTU>mtu_enable=0,wan_mtu= 999</ WAN_MTU> <WAN_MTU>wan_mtu=777</WAN_ MTU> WAN DNS <WAN_DNS>wan_dns</WAN_DNS> To specify one DNS address: <WAN_ DNS>wan_dns=192. 1 68.0.21</WAN_ DNS> <WAN_DNS> wan_dns: DNS IP address; separate multiple addresses with a space To specify multiple DNS addresses: <WAN_DNS>wan_dns=192.
Page 234 Provisioning Reference (WRP400) Feature/XML Parameters Examples Single Port <SINGLE_PORT_FORWARDING>forward To forward FTP to 192. 1 68. 1 5. 1 8: Forwarding _single=name:on|off:both|tcp|udp:external <SINGLE_PORT_FORWARDING>forward_singl -port:internal-port:ip</ e=FTP:on:tcp:21:21:18</SINGLE_ <SINGLE_PORT_ SINGLE_PORT_FORWARDING> PORT_FORWARDING> FORWARDING> NOTE: To configure port forwarding, To configure port forwarding for a non-standard application: <SINGLE_PORT_ you also should configure a DHCP FORWARDING>forward_single=fw1:on:...
Page 235 Provisioning Reference (WRP400) Feature/XML Parameters Examples Port Range <PORT_RANGE_FORWARDING>forward To allow forwarding on two specified port Forwarding _single=name:on|off:both|tcp|udp:port ranges: <PORT_RANGE_FORWARDING> range start:port range end:ip</ forward_port=prf1:on:tcp:555:666:18 <PORT_RANGE_ PORT_RANGE_FORWARDING> </PORT_RANGE_FORWARDING> FOWARDING> <PORT_RANGE_FORWARDING> NOTE: To configure port forwarding, forward_port=prf2:on:both:777:888:19</PORT_ you also should configure a DHCP RANGE_FORWARDING>...
Page 236 Provisioning Reference (WRP400) Feature/XML Parameters Examples Router Syslog <ROUTER_SYSLOG>log_provision</ To configure console display and system log: ROUTER_SYSLOG> <ROUTER_SYSLOG> log_provision=2</ <ROUTER_SYSLO ROUTER_SYSLOG> G> log_provision: Type of log; 0 (console display), 1 (system log), or 2 (console display and system log) ATA Administration Guide...
Page 237 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/go/smallbiz. NOTE Q. I want to use a different computer to access the administration web server.
Page 238 Troubleshooting 3. Click Tools. Click Internet Options. Click the Security tab. Click the Default level button. Make sure the security level is Medium or lower. Then click the OK button. Q. How do I save my current configuration? A. Currently, the only way is to do HTTPGET from an HTTP client, from which you get the entire HTML page.
Page 239 Troubleshooting Press the appropriate code to reset the unit: • Press 877778# to reset the unit to the defaults as it shipped from the ITSP. This will reset the User account password to the default of blank. • Press 73738# to perform a full reset of unit to the factory default settings. The Admin account password will be reset to the default of blank.
Page 240 Troubleshooting STUN does not work with a symmetric NAT router. Enable debug through NOTE STUN Test Enable syslog (see FAQ#10), and set to yes. The messages indicate whether you have symmetric NAT or not. ATA Administration Guide...
Page 241 Environmental Specifications This appendix provides the specifications for the following ATAs: • “PAP2T,” on page 239 • “SPA2102,” on page 240 • “SPA3102,” on page 240 • “SPA8000,” on page 241 • “WRP400,” on page 242 • “WRTP54G,” on page 242 PAP2T Device 3.98”...
Page 242 Environmental Specifications SPA2102 Storage 10% to 90% relative humidity, Non-Condensing Humidity 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 Certificatio FCC (Part 15 Class B), CE, ICES-003...
Page 243 Environmental Specifications SPA8000 Certificatio FCC (Part 15 Class B), CE, ICES-003, A-Tick Certification, RoH Operating 32º to 113º F(0 to 45ºC) Temp Storage -13º to 185ºF (-25 to 85ºC) Temp Operating 10% to 90% relative humidity, Non-Condensing Humidity Storage 10% to 90% relative humidity, Non-Condensing Humidity SPA8000 Device...
Page 244 Environmental Specifications WRP400 WRP400 Device 5.51” x 5.51” x 1.06” (140 x 140 x 27 mm) Dimensions Unit Weight 10.05 oz (285 g) Power External, Switching 5VDC 2A Certificatio FCC (Part 15 Class B), CE, ICES-003, RoHS, UL, A-Tick, NZ Telepermit, CB, Wi-Fi (802.
Page 245 Environmental Specifications WRTP54G Operating 10% to 85% relative humidity, Non-Condensing Humidity Storage 5% to 90% relative humidity, Non-Condensing Humidity ATA Administration Guide...
Page 246 If you use an older web browser, you may have to add http:// in front of the web address. Resource Link www.cisco.com/web/partners/sell/smb/ Cisco Partner Central (requires partner registration and login) Cisco Small Medium www.cisco.com/go/smallbiz...
Page 247 Where to Go From Here Related Documentation 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. These documents and more are available at www.cisco.com/go/smallbiz.
Page 248 Guide users • Phone features • SPA9x2 series IP phones • Administration and use Analog Telephone VARs, system of Cisco Small Adapter Administration administrators, and Business ATAs Guide Service Providers • PAP2T, SPA2102, SPA3102, SPA8000, WRP400, and WRTP54G User Guide for switch...
Page 249 Additional Information This appendix provides links to resources that provide additional information about Cisco Small Business and Cisco Small Business Pro products and services. Resource Location End User License Agreement www.cisco.com/go/smallbiz Regulatory Compliance and www.cisco.com/go/smallbiz Safety Information Warranty Information www.cisco.com/go/smallbiz Cisco Partner Central site for www.cisco.com/web/partners/sell/smb/...
Page 250 Support Contacts To obtain current support contact information for Cisco Small Business and Small Business Pro products, visit the following URL: www.cisco.com/go/smallbiz ATA Administration Guide...

This manual is also suitable for:

Small business pro spa2102 Small business pro spa8000 Small business pro papt Small business pro wrp400

Cisco Small Business Pro SPA3102 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

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Summary of Contents for Cisco Small Business Pro SPA3102

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Table of Contents